Input level adjust system and method

ABSTRACT

A method of matching input amplitudes in a system wherein one or more of a plurality of inputs may be selected, with each input capable of having different characteristics, involving selecting an input signal and mapping the input signal to a predetermined signal amplitude through the use of level matching logic. The level matching logic may include a gain cell for increasing or decreasing the amplitude of the input signal.

RELATED APPLICATION

[0001] This application is related to Provisional U.S. PatentApplication No. 60/167,944, filed Nov. 29,1999, and Provisional USPatent Application No. 60/236,397, filed Sep. 28, 2000, entitledPARTITIONED SIGNAL PROCESSING SYSTEM WITH AUTOMATIC NOISE COMPENSATIONAND METHOD, and having the same inventors as the present application.

FIELD OF THE INVENTION

[0002] This invention relates to the art of distributed signalprocessing systems that may include companders, noise compensators, andmethods of controlling systems that include companders, volume controlsand noise compensators.

BACKGROUND OF THE INVENTION

[0003] Existing audio signal processing systems suffer from a variety oflimitations. Some of the limitations are imposed by the transmission orstorage medium or other technological deficiencies; other limitationsare the result of environmental issues. Regardless of the reason, theresult is the same: the listener receives a less than optimal listeningexperience.

[0004] For example, the amplification necessary to hear the quietestportions of an audio signal may result in maximum amplitudes that areundesirably loud. Conversely, amplitudes allowing loud portions of anaudio signal to be heard at a comfortable level, may result in not beingable to hear quiet portions of the signal. Enabling the entire signal tobe comfortably heard at all times by the listener requires that theinput source dynamic range of the signal be transformed into the dynamicrange of the listener's environment and ability. Companders aresometimes used to correct the problem of inadequate dynamic rangetransformation. In many situations, the dynamic amplitude range of asignal exceeds the capabilities of its transmission channel, receiver,or restrictions of its environment. These limitations make it desirableto compress the dynamic amplitude range of the signal to allow allportions of the signal to be discerned.

[0005] Shown in FIG. 1A is a representative embodiment of a prior artcompander, which receives an input signal in both a power estimatorcircuit and gain multiplier. The typical prior art power estimatorprovides a linear output to a gain calculate circuit. A control signalmay also be provided to the power estimator circuit, typically to modifythe attack or release characteristics of the estimator, and the gaincalculate circuit, typically to change the amount of compression orexpansion. The output of the gain calculate circuit is then combined inthe gain multiplier with the input signal to provide a companded signal,taken as an output. A typical power estimator is implemented as a peakdetector, for example as shown in FIG. 1B. In FIG. 1B, the input signalis provided to a diode. The output of the diode is tied to groundthrough an RC circuit, with the output taken at the node connecting thediode, resistor and capacitor. The capacitor charges up to the peakinput voltage level and is gradually reduced over time by the resistor.Alternatively, an integrator circuit, also know as a low pass filter,may be used as a power estimator, as shown in FIG. 1C, where the inputsignal is provided to a conventional RC integrator. Peak detectors andlow pass filters suffer from increasing low frequency distortion as thesignal frequency approaches the corner frequency of the circuits.Lowering the corner frequency to decrease the low frequency distortionincreases the transient response time leading to overamplification andsignal clipping and underamplification. Dividing the total bandwidthinto multiple frequency bands and using multiple companders can reducedistortion and transient response time but at the cost of the extraprocessing for the additional companders. A typical prior art stereocompander is shown in FIG. 1D, with left and right input channels eachsupplied to a multiplier and multiplier value, typically a value of ½.The multiplier outputs are then added and supplied to a pair ofconventional, prior art companders. The multipliers and adder form aninput signal mixer and is used to maintain relative spatial informationin the two channels. The right and left channel input signals are alsosupplied to the respective companders, with the output of the respectivecompanders being available as the output signal. The input signal mixingtypically results in additional distortion since the adjacent channelsignal is partly controlling the compander gain. It also typicallyresults in one channel being under-amplified and the other channel beingover-amplified which can result in clipping distortion.

[0006] Broadcast or recording restrictions or other technologicallimitations often mandate that the maximum amplitude range of a signalbe restricted, and as a result signals are compressed to remain withinthose restrictions and limitations. After such a signal has beenreceived or recovered, it is often desirable to expand its amplituderange to restore the original dynamic amplitude range. Thus a compressorand expander pair can be used to cancel out low frequency distortion,but do not and cannot function as a standalone compressor or expander.

[0007] Referring to FIG. 1E, a prior art compander pair may be betterappreciated, where an input signal is provided to a first compander forpurposes of compressing the input signal. The compressed signal, whichincludes some distortion, is then recorded on a suitable media, orotherwise managed. The compressed signal, with distortion, is thereafterprovided a second compander in the pair. The second compander expandsthe compressed signal, and removes the distortion, resulting inrestoration of the input signal, which is then provided as the systemoutput signal.

[0008] One common limitation of prior art audio signal processingsystems is that it is often difficult for a listener to comfortably hearall portions of an audio signal due to environmental audio noise. Insuch circumstances, when the amplification is sufficient that the louderportions can be discerned easily, the quiet (or low amplitude) portionsof the signal are masked by the environmental noise. Environmental noiseis transient and often unpredictable in nature, which makes manualadjustment to compensate for it particularly difficult. If the usermanually increases the volume for times when the signal cannot be hearddue to loud transient noise, the volume will be too loud once when thenoise has subsided or the audio signal becomes greater in amplitude.

[0009] This problem occurs in many situations, usually (though notalways) involving outdoors or mobile environments such as a car stereo,a cellular telephone used in public places, a portable radio used duringa public sporting event, or home theater systems. There are also devicessuch as alarms, door bells, and phone ringers, that cannot be heard attimes when there is much environmental noise, or must be setuncomfortably loud in order that the listener is assured of hearingthem.

[0010] There have been many attempts at solutions to the problem ofinadequate noise compensation, a representative embodiment of a priorart solution being shown in FIG. 2.—The environmental noise signal iscalculated as the difference between the environmental input (noise plusspeaker output) detected by a microphone and a representation of thesignal output by the speaker (a.k.a. reference). This environmentalnoise signal is then provided to a power estimator, typically anintegrator or lowpass filter to smooth the signal. The power estimatoroutput is provided to a gain calculate circuit to calculate a gain valueto increase the output level of the input signal, typically in a linearmanner. This type of prior art solutions provides no means ofcalibration of the circuitry to the acoustic environment making theirproper operation unpredictable. They also do not satisfactorily addressproblems with variations in the audio source such as long silent pausesin the audio signal, uncontrolled positive feedback known as gain chase,room acoustic resonances that appear as false noise, or allow changingthe minimum signal to noise ratio of the system. Further, they have aninadequate response to noise in that some respond too quickly, reactingto phone ringers and short bursts of speech, while others have too longand inaccurate a response. Prior art solutions also do not allow theuser to select the signal or noise priority so that if the noise isspeech, the signal source will be reduced instead of increased.

[0011] Further, when sound is to be heard in multiple locations, eachlocation has a different environmental requirement. The signal necessaryto provide adequate sound in one location, often results in sounds thatare either too loud or quiet in other areas. This problem is typicallysolved by the use of redundant equipment, for example one prior artaudio system per room.

[0012] It is also desirable to have methods to allow a user to perform anumber of signal adjustments, e.g. calibration, changing compression orexpansion factors, changing the minimum signal to noise ratio, channelbalancing and equalization, in order to obtain optimal sound for a givenenvironment. It is even more desirable to have these adjustments doneautomatically since most users lack the knowledge and understanding ofhow to perform correct and optimal adjustments to their equipment. Priorart solutions have no automatic adjustments and require manualadjustments with little or no explanation. Several techniques have beendeveloped to address some of these problems.

[0013] Blackmer U.S. Pat. No. 3,789,143 (1974) discloses a companderthat performs a logarithmic transformation of a signal, proportioned tothe root mean square of two 90-degree phase separated signals. Whilethis technique will work for any particular frequency, it is not able tomaintain a 90-degree separation over all frequencies.

[0014] Beard U.S. Pat. No. 4,169,219 (1979) teaches the use of ananalog, low-pass-filter delay-buffer in implementing acompressor-expansion pair, for recording a compressed signal thatsubsequently would be expanded upon being played. It is unsuitable forstandalone use due to significant distortion of low frequency signals.

[0015] Bethards U.S. Pat. No. 4,216,427 (1980) instructs in the use ofan adaptive audio compressor using analog techniques for restrictingsubsequent RF signal modulation. Changes in gain occur continuously,resulting in significant distortion of low frequency signals.

[0016] Orban U.S. Pat. No. 4,249,042 (1981) shows the use of amulti-frequency band compressor that controls the gain in each band bymeans of measuring the power in a master band. It has no mechanism toavoid amplifying the noise floor, and no dynamic input signal mapping.

[0017] Schroder U.S. Pat. No. 4,306,202 (1981) discloses a discretecompressor and expander, implemented using analog techniques. Thecompressor and expander modes, selected by analog switches, are intendedto be used only in combination with each other. The compressor cannot beused in standalone operation because of distortion of low frequencysignals.

[0018] Bloy U.S. Pat. No. 4,368,435 (1983) shows the use of a narrowband compander with a combination of a fast and slow attack circuit. Bydividing the input into multiple narrow frequency bands, they attempt tominimize signal distortion. This technique provides unacceptabledistortion of wide band audio signals.

[0019] Unagami et al. U.S. Pat. No. 4,482,973 et al. (1984) is not acompander per se, but instead teaches the use of two AGC (automatic gaincontrol) circuits. Its algorithm does permit its implementation in a DSPand it performs a signal limiting function. The fixed delay causes bothsynchronization and low frequency response problems.

[0020] Stikvoort U.S. Pat. No. 4,562,591 (1985) discloses the use of apeak detector with a non-linear amplifier to provide a compander. Itsuse of a low-pass filter results in high distortion at low frequencies,and its overall design causes undesirable signal clipping.

[0021] Rosback U.S. Pat. No. 4,641,361 (1987) instructs in the use of ananalog, multi-frequency band, automatic gain circuit that makes use of apeak clipper to reduce overall gain. Gain is changed continuously,resulting in distortion at low frequencies.

[0022] Bloy et al. U.S. Pat. No. 4,853,963 (1989) shows the use of a DSPto process narrow band signals. Its algorithm causes high levels ofdistortion on wide band signals.

[0023] Jorgensen U.S. Pat. No. 4,859,964 (1989) teaches the continualupward and downward adjustment of an automatic gain control to keep aninput signal within certain limits. This technique undesirably amplifiesthe signal noise floor. It does not disclose the algorithms used in themicroprocessor.

[0024] Thomas U.S. Pat. No. 4,947,133 (1990) instructs in the use of acompressor that uses a fixed signal delay line. After a signal zerocrossing occurs, the compressor performs various signal smoothing, gain,sample and hold and compression functions. It performs no signalexpansion and is implemented using a combination of analog and digitalcircuitry.

[0025] Akagiri et al. U.S. Pat. No. 4,972,164 (1990) teaches thespecific design of a curvilinear compander. It is a complex design thatuses both analog and digital circuitry. Its curvilinear algorithmminimizes distortions and abrupt transitions of the companded signalover its entire input range.

[0026] Orban U.S. Pat. No. 5,444,788 (1995) shows the use of an analogcompander. Its use of diodes in a non-linear low-pass filter causestemperature stability problems and distorted low frequency response.

[0027] Werrbach U.S. Pat. No. 5,463,695 (1995) instructs in the use ofanalog tracking filters to implement a compressor. It performsnon-linear compression that compresses transient peaks more than averagesignals. The average compression is fixed and it does not perform anydynamic range mapping.

[0028] Frey et al. U.S. Pat. No. 5,631,968 (1997) discloses an analogdesign with a variable compression ratio controlled by the time-averagedaudio signal and various breakpoints. Low, selected and high compressionratios are used depending on the time-averaged signal. Changes in gainare made continuously, resulting in inherent signal distortion.

[0029] U.S. Pat. No. 4,322,579 to Kleis et al. (1982) disclosesdetecting the environmental noise level. It starts compressing thesignal when a certain threshold is reached. The use of a band-pass andhigh-pass filter results in a substantial reduction in signal fidelity.

[0030] U.S. Pat. No. 4,553,257 to Mori et al. (1985) describes an openloop automatic volume control device. It is a single channel, analogcircuit that does not alter the signal if the noise is below a giventhreshold. It performs some amount of variable compression as a functionof signal and environmental noise above a given threshold. The primarydisadvantage of this invention is that it suffers from positive feedbackbetween the speaker and microphone, making it only useful for headphoneapplications. It provides no limit on the maximum volume produced.

[0031] U.S. Pat. No. 4,628,526 to Germer (1986) teaches using the rateof change of ambient noise and signal, to determine how the signalshould be adjusted. If the noise is increasing faster than the signal,it increases the signal amplification, if the signal is increasingfaster than the noise, it decreases the signal amplification. It has thedesirable feature of not needing user calibration. It has thedisadvantage that it always amplifies the signal, when compressing thesignal would provide superior comfort to the user. It also requiresadditional circuitry to address the problem of dealing with silentportions of the audio signal.

[0032] U.S. Pat. No. 4,868,881 to Zwicker et al. (1989) shows using amicrophone to detect noise. A multi-band equalizer is used to processthe noise and audio signals, and the resultant composite signal isamplified and fed back into the noise compensator circuit. Thedifficulty with this patent is that it requires that the microphone onlydetect noise, and thus places it in a vehicle engine compartment, whenin fact, wind noise is often the dominant source of environmental noise,resulting in the circuit not solving the stated problem.

[0033] U.S. Pat. No. 4,882,762 to Waldhaner (1989) teaches the use of aprogrammable multi-band compression system for hearing aids. It providesdifferent amounts of fixed compression for multiple frequency bands,compensating for hearing loss that is both audio amplitude and frequencydependent. This patent addresses the issue of compressing a signal tocompensate for hearing loss. It does not provide variable, automaticcompensation of the signal in the presence of environmental noise, sinceit has no way to detect or distinguish this noise.

[0034] U.S. Pat. No. 4,891,837 to Walker et al. (1990) discloses thecompression or expansion of a signal for use in a speakerphone. Theamount of signal transformation is a function of the ambient noise. Theprimary source of noise addressed by this invention is the signalreceived by the microphone from the speakerphone's speaker during aduplex conversation. The invention assumes that the user will speaklouder in a room with high ambient noise, and compensates accordingly.

[0035] In U.S. Pat. No. 4,953,221 to Holly et al. (1990) shows howpositive feedback problems can be avoided by converting noise and audiosignals to DC levels and subtracting them from each other. Thedisadvantage of this technique is that a sample and hold circuit must beused to avoid a noise problem whenever the audio input signal goessilent.

[0036] U.S. Pat. No. 5,107,539 to Kato et al. (1992) discloses the meansfor adjusting the surround or effect sound in a vehicle as environmentalnoise is sensed. It amplifies, as a function of the sensed noise, thesignal and the effect transformation of the signal, using unique levelcontrol circuits for each signal. This circuit has severaldisadvantages, long silent pauses in the audio signal will result inundesirably loud amplification, and it does no compression of thesignal, resulting in situations where the audio signal becomes too loud.

[0037] U.S. Pat. No. 5,172,358 to Kimura (1992) shows the usage of adigital signal processor to boost low and high frequencies depending onthe average amount of sound pressure. No means are provided to calibratethe actual sound pressure with the levels inside the circuit. Thecontroller is the means for controlling the device, yet no algorithmsare disclosed for implementing it. There is no mechanism disclosed forhandling signals that rapidly increase or decrease.

[0038] U.S. Pat. Nos. 5,434,922 and 5,615,270 to Miller et al.(1995/1997) teaches the use of adaptive algorithms with a digital signalprocessor to determine the amount of noise and dynamically compensatefor it. Adaptive and least means square algorithms are computationallyintensive and in certain situations can add undesirable amounts ofdistortion to a signal. The instruction is unclear as how signalprocessing is performed; FIG. 9 of this patent implies that theinvention only performs fixed, 2:1 compression; how to set the minimumlimit in item 62; or how to set the compression parameters in the gaincalculator shown in item 60. Attack and release are fixed. Usage is madeof prior art buffers to perform filter delay compensation. No provisionis made to squelch inherent signal source noise. Compression occurs evenwhen there is no environmental noise.

[0039] U.S. Pat. No. 5,450,494 to Okubo et al. (1995) shows the use ofadaptive filters with a digital signal processor to determine the amountof noise and dynamically compensate for it. Their invention makes use ofa fast Fourier transform which is computationally expensive, todetermine the coefficients for the adaptive filters. It assumes thatnoise has a fixed frequency spectrum, dominant in lower frequencies andattenuated at higher frequencies. It teaches the theory of soundpressure and noise.

[0040] U.S. Pat. No. 5,509,081 to Kuusama (1996) teaches the use ofselective amplification of various frequency bands in order to maskunwanted noise. No signal companding is performed, resulting incircumstances where certain noise-dominant frequency bands are amplifiedpainfully loud. The main distinctive feature is the use of a delay linethat is ineffective due to room reverberation and variations caused bychanging microphone position. This method works for fixed delays, butdoes not function well in situations where there is considerable phasedelay, signal dispersion or echoes. It provides no instruction on how tocalibrate the circuitry.

[0041] U.S. Pat. No. 5,530,761 to d'Alayer de Costemore d'Arc (1996)teaches the use of a mathematical algorithm for automatically adjustingsound volume. It performs no signal companding and does not appear tohave addressed considerations such as avoidance of gain chase andcalibration.

[0042] U.S. Pat. No. 5,550,922 to Becker (1996) discloses the use of ananalog compressor. It attempts to avoid a gain chase problem by matchingthe output signal to exceed the environmental noise by a small margin.It provides the means to reduce the gain during signals below aparticular threshold. It provides no information on how to measurenoise, set the volume or perform calibration.

[0043] U.S. Pat. No. 5,666,426 to Helms (1997) discloses the use of adigital signal processing algorithm to provide automatic volume controlby maintaining a constant signal to noise ratio. The system calibratesitself by sensing the ambient sound level shortly after being poweredon. No signal companding is performed, resulting in circumstances wherethe output volume is unacceptably loud. It provides instruction on avolume control and calibration, but requires the room to be quiet whencalibrating. It does not address normal mode resonances due to roomacoustics providing incorrect calibration.

[0044] U.S. Pat. No. 4558460 to Tanaka, et al. (1985) describes a motorvehicle speed sensor used to increase the output of an amplifier. Thiswill not work in a non-automotive setting since the noise compensationis dependent on vehicle speed and not environmental noise.

[0045] As will be appreciated from the foregoing discussion, the priorart companders suffer from a number of disadvantages. All produce highlevels of signal distortion at low frequencies. Using multiple frequencyband compander techniques to reduce distortion requires significantadditional processing requirements. Many rely upon a compressor andexpander pair to cancel out low frequency distortion, and do not andcannot function as a standalone compressor or expander. Many of thecompanders are implemented using analog designs that do not addressconsiderations necessary for the use of digital signal processors, ortake advantage of their capabilities. Some use multiple channelcompander designs with input signal mixing that result in inter-channelmodulation distortion and output clipping. Most compander designs usefixed attack and release times or use a plurality of fixed attack andrelease time-constant filters, that can result in additional signaldistortion or limited operating range. Further, prior art noisecompensators suffer from gain chase problems, inadequate and inaccurateresponse to noise, a lack of signal or noise priority choice, and nomeans to vary the output signal to noise ratio. Prior art systemstypically require redundant equipment and do not provide automaticadjustments.

[0046] In summary, existing inventions are ineffective due to inherentdesign limitations.

SUMMARY OF THE INVENTION

[0047] The partitioned signal processing system and method of thepresent invention substantially overcomes each of the aforementionedlimitations of the prior art. Because of the substantial flexibility ofthe system and method of the present invention, cost effectiveimplementations may be envisioned which range from alarms, ringers andphones, to car stereos and home entertainment systems, to recordingstudios and theaters, and to applications distributed across a network.In addition, the system and method are capable of use in other non-audiosignal processing applications. A feature of many implementations of theinvention is that substantially arbitrary, dynamic, and flexibleprocessing throughout an environment becomes possible. The architectureof the present invention permits both digital and analogimplementations, and also permits partitioning of both hardware andsoftware. Moreover, modules can be concentrated locally, or dispersedacross a network which may include communications links. In at leastsome embodiments it may be desirable to permit reconfiguration of themodules; this can be accomplished either through software such asdynamic switching or packet routing, or through hardware such ascrossbar switches or remote controls.

[0048] The architecture of the present invention can be divided into aplurality of modules, which may also be characterized by the processingthey provide, including the following:

[0049] User Interface—The user interface provides a means for theoutside world to interact and control the partitioned signal processingsystem. Inputs can be from traditional keyboards and remote controls orvia remote links to offsite facilities such as customer service centersfor remote diagnostics or internet websites. Outputs from the systeminclude the remote links and displays to show information to the userregarding the system, its configuration, control, and status.Implementation of the displays can range from discrete LEDs, to bargraphs, to bit mapped displays. Outputs to the system include Operation,Set-up, and Configuration commands. Operation commands include thetypical volume control, input source selection, tone controls, station,disk and track selection. Also included are preference controls such asnoise sensitivity, noise reaction time, and noise compensation priority.The noise compensation may be set to give priority to the audio signalover the environment (noise compensation), such as might occur while afamily is watching a video, or to cause the audio signal to be secondaryto the environment (signal muting), such as during a conversation.Set-up commands typically include calibrate (balance) the system toallow noise compensation, and set minimum and maximum volume levels toenable dynamic range mapping. Configuration commands allow a multitudeof different system module configurations to be implemented out of asystem's resources which may be increased or decreased over time. Thisfunction need not be provided if a fixed module implementation willalways be used. Various configurations can be implemented in a varietyof manners, ranging from dynamic software module selection, to crossbarswitches, to assignment of packet routing, to usage of remote controlsto select module function. Further, the statistical analysis engines maybe used either to initiate automatically certain commands and actions,or to provide information to assist an operator in optimizing systemoperation. An annunciator may be used in some embodiments to providesynthesized spoken advice to the user, advising them of a necessary orsuggested action for a particular procedure.

[0050] Input Signal Preprocessor—The input signal preprocessor allows amultitude of input signals to be multiplexed, synchronization ofasynchronous events and data, and other signal processing such as analogto digital data conversion, bandsplit filtering and equalization, andinput level adjusting to be performed. The input level adjust functionadjusts the peak input signal (largest amplitude input signal) suppliedby the selected source to the 0-dB level (largest amplitude signal)allowed by subsequent processing modules and allows for optimumcompander operation. Bandsplit filtering and equalization allowsdividing a signal into a multitude of frequency bands, and thenperforming equalization on each frequency band as required and allowsfor multiband signal processing.

[0051] Adaptive Dynamic Compander—The Adaptive Dynamic Compander allowsextremely low signal distortion, wide bandwidth, dynamic range mappingby matching the input source dynamic range to the listener's dynamicrange by either compressing or expanding an input signal. Extremely lowdistortion is provided by the half-wave signal processor which causeschanges in gain only at zero crossings to minimize signal distortion.The half-wave signal processor may also include an optional synchronizerto insure that the calculated gain is used on the correct half-cycle,leading to improved transient response, a variable attack and releasefeature (also used in other modules) to dynamically modify the companderresponse time to varying input levels, a post-power estimator mixer thatallows multiple companders to be used in multiple channel/bandconfigurations, without altering relative spatial information or causingsignal distortion, and a gain calculator for determining the amount ofinstantaneous gain by which a signal should be multiplied to becorrectly companded. Many audio signals consist of a plurality ofchannels used to provide stereo or theater sound reproduction. Whencompanding audio signals, it is desirable to preserve the relativeamplitudes of the channels to each other, to preserve spatial locationinformation. Using a post-power estimator mixer in each compander or ina centralized power estimator mixer, allows uniform changes to be madeon all channels, preserving spatial signal intensity information, andminimizing inter-channel/band distortions and clipping. The gaincalculator may include a look-ahead output clip detector for minimizingoutput signal clipping distortion for signals changing faster than thereaction time of the adaptive dynamic compander. Alternatively, a softclip function may be used for minimizing output signal clippingdistortion.

[0052] Central Post Power Estimator Mixer—Use of a Central Post PowerEstimator Mixer reduces processing requirements and allows multiplecompanders to be used in multiple channel/band configurations withoutaltering relative spatial information or causing signal distortion orclipping.

[0053] Volume Control and Pre-Mixer—The Volume Control and Pre-Mixerselects either input signals or a calibration signal to be output andallows the user to make fine adjustments of the output signal amplitude.A pre-mix function allows a multitude of input signals, typicallymultiple frequency bands or channels, to be combined into a given volumecontrol.

[0054] Output Signal Processor—The Output Signal Processor is used togenerate multiple band outputs for speaker equalization andmulti-amplification implementations, performs data conversion on thesignal to be converted into sound and generates reference signals usedin noise compensation. Bandsplit filtering, equalization, signal mixing,soft clip functions, and amplification may also be performed. Analog anddigital outputs may be produced.

[0055] Noise Extractor—The Noise Extractor of the present inventiontypically comprises a loop processor and a noise processor, and may beimplemented in any of a plurality of configurations, including closedloop, open loop, or a combination of both open and closed loops alsoreferred to as a leakage loop. The loop processor compares the estimatedpower of the signals detected by the environmental sensors, typicallymicrophones or other sensors, with the reference power estimates of theoutput signal processor (signals prior to speaker amplification), andproduces a noise signal indicating the level of environmental noise.Various optimizations may be provided for each configuration, includingacoustic loop balancing for calibrating the system and negative feedbackloops which are used to eliminate the “gain chase” runaway volumeincrease problem by compensating for inaccurate calibration and changesin the listening environment, e.g. people and objects moving in theroom, curtains being opened or closed, room resonances and otherperturbations. The environmental noise signal is further processed bythe noise processor. The noise processor may provide corrections to theenvironmental noise signal caused by the negative loops, a sensitivitycontrol for setting or modifying the output signal relative to theenvironmental noise level, and/or a variable attack and releaseprocessor to provide a selective response feature for permittingaccurate and quick response to noise sources having a duration whichexceeds a predetermined threshold while at the same time ignoringtransient or short duration environmental noise. The processedenvironmental noise signals may then be used by the other modules,particularly the transform engines, to form positive loops that willincrease or decrease the output signal relative to the environmentalnoise depending on if noise compensation or signal muting mode isactive.

[0056] Transform Engine—The Transform Engine processes user controls,statistics engine outputs, and the environmental noise signals, andprovides the control signals necessary for the proper operations of thesystem. The primary function of the transform engine controller is totransform the user control signals, statistics engine outputs, and theenvironmental noise signals into the control signals necessary forproper operation of each module. It continually determines how much thecompander should expand or compress the input signal and the outputlevel of volume control modules.

[0057] Statistical Analysis Engine—The Statistical Analysis Enginesallow for dynamically monitoring a multitude of signals, and allowingboth automatic and manual control, optimization, and modification ofoperating parameters of the modules within the system based upon thisinformation. The statistical analysis module is composed of a pluralityof histogram generators, which create a histogram of a particularsignal, and statistical analyzers to inspect the histograms and produceflags and data that may be used by other modules to performautomatically certain actions, or to provide information to assist anoperator in optimizing system operation.

[0058] Calibrator/Annunciator—A calibrator produces signals for use insetting, adjusting and calibrating the system. Typical uses includesetting the minimum and maximum volume levels, environmental noisecompensation loop balancing calibration and output signal equalization.Calibration signals are typically produced by white or pink noisegenerators, random number generators, arbitrary waveform generators,Fourier synthesis, amplitude, frequency, and phase modulators, andanalog waveform generators. An annunciator may be included in someembodiments to provide spoken advice to the user, advising them of anecessary or suggested action for a particular procedure or informingthe user of the status of the system. This information can beimplemented by means of voice synthesizers, voice compression, or otherprior art techniques.

[0059] The methods of the present invention can be divided into manualand automatic methods, which may also be characterized by the processingthey provide, including the following:

[0060] Manual Methods to Establish Dynamic Range Mapping—Methods usingthe previously described modules to set the compander kneepoints,companding ratio, and system gain by manually setting the maximum,minimum, and/or typical listening levels of a partitioned signalprocessing system are disclosed. These levels are used to determine thelistening output dynamic range and establish dynamic range mapping fromthe source input dynamic range to the listening output dynamic range.

[0061] Automatic Methods to Establish Dynamic Range Mapping—Methodsusing the previously described modules to establish the maximum andminimum listening levels and resulting listening output dynamic range ofa partitioned signal processing system from a default set of companderkneepoints, companding ratios, system gain, levels and thresholds aredisclosed.

[0062] Automatic Method to Maintain Dynamic Range Mapping—A method usingthe previously described modules to maintain the minimum listening levelof a partitioned signal processing system by modifying companderkneepoints, companding ratios, and system gain while the user modifiesthe volume control setting is disclosed.

[0063] Automatic Method to Adjust Dynamic Range Mapping in the Presenceof Noise—A method using the previously described modules to adjust theminimum listening level and /or volume control levels of a partitionedsignal processing system by modifying compander kneepoints, compandingratios, and/or system gain to compensate for environmental noise isdisclosed.

[0064] “Intelligent Volume Control” Automatic Method—A method using thepreviously described modules to respond to user control adjustments,typically the volume control, in the absence or presence ofenvironmental noise, to appropriately adjust the maximum and minimumlistening levels by modifying compander kneepoints companding ratios,system gain, and noise sensitivity of a partitioned signal processingsystem is disclosed. This method allows for one simple user control tomodify the dynamic range mapping of the system to provide optimallistening in any acoustic environment.

[0065] The foregoing features of the invention, including additionalprocess, system, apparatus and method aspects, may be better appreciatedfrom the following Detailed Description of the Invention, taken togetherwith the attached Figures.

THE FIGURES

[0066] FIGS. 1A-1F show prior art signal processing systems.

[0067]FIG. 2 shows a prior art example of a noise compensation system.

[0068]FIG. 3A shows a top level configuration of a partitioned audiosystem in accordance with the present invention.

[0069]FIG. 3B shows a top level process flow diagram in accordance withthe present invention.

[0070]FIG. 4 shows a partitioned signal processing system in accordancewith the present invention.

[0071]FIG. 5A shows an exemplary home-wide audio server network inaccordance with the present invention.

[0072]FIG. 5B shows an audio visual file server in accordance with thepresent invention.

[0073]FIG. 5C shows an exemplary centralized partitioned signalprocessing system.

[0074]FIG. 5D shows an exemplary system for a single listeningenvironment, with noise compensation.

[0075]FIG. 5E shows a exemplary system for a single listeningenvironment with local I/O, but without noise compensation.

[0076]FIG. 5F shows an exemplary embodiment of an intelligent speaker inaccordance with the present invention.

[0077]FIG. 5G shows an exemplary implementation of the present inventionas a hearing aid.

[0078]FIG. 5H shows an exemplary stereo system implementation.

[0079]FIG. 5I shows an exemplary eight channel studio mixer inaccordance with the present invention.

[0080]FIG. 6A shows in block diagram form a user interface in accordancewith the invention.

[0081]FIG. 6B shows a process flow diagram for a user interface inaccordance with the invention.

[0082]FIG. 6C shows a more detailed process flow diagram for a userinterface.

[0083]FIG. 7 shows a Pre-Companderflow diagram.

[0084]FIG. 8A shows an Input Signal Pre-Processing Block Diagram.

[0085]FIG. 8B shows the mapping of an input signal to digital space foruse establishing a system signal level, or 0 dB level.

[0086]FIG. 8C shows in block diagram form an exemplary form of inputlevel adjust logic.

[0087]FIG. 8D shows an exemplary bandsplit filters/scaling processor.

[0088]FIG. 8E shows in block diagram form the logic for a digital signalprocessor and gain cell.

[0089]FIG. 8F shows a hybrid signal processor and gain cell blockdiagram.

[0090]FIG. 8G shows an analog signal processor and gain cell.

[0091]FIG. 9A shows the logic for an input version of a clip detector.

[0092]FIG. 9B shows the logic for an output version of a clip detector.

[0093]FIG. 9C shows in block diagram form the logic for a clip detectoranalyzer.

[0094]FIG. 10A shows in flow diagram form the operation of an inputsignal processor.

[0095]FIG. 10B shows in flow diagram form a single loop input leveladjust process.

[0096]FIG. 10C shows in flow diagram form a two loop input level adjustprocess.

[0097]FIG. 10D shows in flow diagram form the Apply BandsplitFilter/Equalizer process.

[0098]FIG. 10E shows a Bandsplit Filterflow diagram.

[0099]FIG. 11 shows an Environmental Sensor Adjustment and NoiseExtractor Block Diagram.

[0100]FIG. 12 shows an Environmental Sensors Adjustment and NoiseExtractorflow diagram.

[0101]FIG. 13 shows a Statistical Engine Block Diagram.

[0102]FIG. 14 shows a Statistical Engineflow diagram.

[0103]FIG. 15 shows a Transform Engine Block Diagram.

[0104]FIG. 16 shows a Transform Engine Example.

[0105]FIG. 17 shows a Transform Engineflow diagram.

[0106]FIG. 18 shows a Centralized Multiple Band/Channel Power EstimatorMixer.

[0107]FIG. 19 shows a Central Power Estimatorsflow diagram.

[0108]FIG. 20A shows an Independent Adaptive Dynamic Compander Group.

[0109]FIG. 20B shows an Adaptive Dynamic Compander Group Block Diagram.

[0110]FIG. 20C shows an Adaptive Dynamic Compander with Low Distortion.

[0111]FIG. 21A shows an example of the Integration of a SynchronizerBlock with a Half-Wave Signal Processor.

[0112]FIG. 21B shows a Synchronizer Input Signal Processing Example.

[0113]FIG. 21C shows a Synchronizer Output Signal Processing Example.

[0114]FIG. 22 shows a Half-Wave Signal Processor Block Diagram.

[0115]FIG. 23A shows an exemplary Embodiment of Half-Cycle and InitialPower Estimators.

[0116]FIG. 23B shows a Changing K and K′ Lowpass Filter Coefficients toCompensate for Changing Effective Sample Rate (Fseff) Resulting inConstant Initial Power Estimator Low Pass Filter Response (Fc).

[0117]FIG. 23C shows exemplary Half-Wave Power Estimator Signals.

[0118]FIG. 24A shows a Generic Variable Attack/Release Block Diagram.

[0119]FIG. 24B shows a block diagram of a One K″ Segment VariableAttack/Release Processor example.

[0120]FIG. 24C shows an exemplary Embodiment of an Attack/ReleaseProcessor Using a Single Segment Nonlinear Adjuster Coefficient K″.

[0121]FIG. 24D shows a Fixed Tracking Adjuster Filter Coefficient K″Graph.

[0122]FIG. 24E shows a One Segment Linear Variable Attack/Release GraphUsing K″=B*Δ+A Segment Processing Transform.

[0123]FIG. 24F shows a One Segment Nonlinear Variable Attack/ReleaseGraph Using K″=A*ΔA**2+B*Δ+C Segment Processing Transform.

[0124]FIG. 24G shows examples of Intermediate Power Estimates forCompander Use.

[0125]FIG. 25A shows a Local Post Power Estimator Mixer Found in EachCompander.

[0126]FIG. 25B shows a Local Post Power Estimator Mixer example withmultiple external inputs.

[0127]FIG. 26A shows a Generic Gain Calculate Block Diagram example of aSegmented Mapping Converter.

[0128]FIG. 26B shows an exemplary Serial Gain Calculate Embodiment withPredictive Clip Detection and Gain Correction.

[0129]FIG. 26C shows a Generic Segmented Gain Calculate Block Diagram.

[0130]FIG. 26D shows a Gain Calculate Example Using Four Segments.

[0131]FIG. 26E shows an Input Power to Gain Transform Graph Using ThreeLogarithmic Segments.

[0132]FIG. 26F shows a Three Segment Input Power to Output Power Graphfor Various Companding Ratios.

[0133]FIG. 26G shows a MX+B Line Gain Transform.

[0134]FIG. 26H shows a Single Segment Curvilinear Gain Calculate BlockDiagram Example.

[0135]FIG. 26I shows a Graph of a Curvilinear Input Power to GainCalculation Example Using Single Segment Non-Linear Computation Actingas Four, Smoothly Connected Pseudo Segments.

[0136]FIG. 26J shows a Curvilinear Input Power to Output Power Graph forVarious Companding Ratios.

[0137]FIG. 27A shows a Companderflow diagram.

[0138]FIG. 27B shows a Split Compander—Part Aflow diagram.

[0139]FIG. 27C shows a Split Compander—Part Bflow diagram.

[0140]FIG. 28 shows a Half-Wave Signal Processor flow diagram.

[0141]FIG. 29 shows a Half Cycle Power Estimate flow diagram.

[0142]FIG. 30 shows an Initial Power Estimators flow diagram.

[0143]FIG. 31 shows an Attack/Release flow diagram.

[0144]FIG. 32 shows a Math Processors flow diagram.

[0145]FIG. 33 shows a Segment Processors flow diagram.

[0146]FIG. 34 shows an Attack/Release Segment Combiner flow diagram.

[0147]FIG. 35 shows a Tracking Adjuster Filter flow diagram.

[0148]FIG. 36 shows a Local Post Power Estimator/Mixer flow diagram.

[0149]FIG. 37A shows a Gain Calculate flow diagram.

[0150]FIG. 37B shows an exemplary Embodiment of Gain Calculate flowdiagram for Parallel Implementation.

[0151]FIG. 37C shows an exemplary Embodiment of an Optimized Two PassGain Calculate flow diagram with Predictive Clip Detection and GainCorrection.

[0152]FIG. 38 shows a Segmented Gain Calculate flow diagram.

[0153]FIG. 39 shows an Update Synchronizer Inputs and Get SynchronizerOutputs flow diagram.

[0154]FIG. 40A shows a Softclip Algorithm flow diagram.

[0155]FIG. 40B shows Soft Clip Waveform Examples.

[0156]FIG. 40C shows a Softclip FIFO Buffer Usage.

[0157]FIG. 41A shows a Volume Control Multi-Module Diagram.

[0158]FIG. 41B shows a Volume Control and Preprocessor Block Diagram.

[0159] FIGS. 42A-D show Volume Control Configurations.

[0160]FIG. 43A shows an Output Signal Processing Block Diagram.

[0161]FIG. 43B shows a Band Group Output Processors Block Diagram.

[0162]FIG. 44 shows an Output Conversions Block Diagram.

[0163] FIGS. 45A-G show Channel/Band Group Processing Configurations.Group Processing ConfigurationsFIG. 46 shows a Volume Control Block flowdiagram.

[0164]FIG. 47 shows a Volume Control and Pre-Processor flow diagram.

[0165]FIG. 48 shows an Output Signal Processors flow diagram.

[0166]FIG. 49 shows an Output Conversions flow diagram.

[0167]FIG. 50 shows a Calibrator/Annunciator block diagram.

[0168]FIG. 51 shows a Conceptual Noise Compensation Loop.

[0169]FIG. 52A shows the Loop Processor Block Diagram.

[0170]FIG. 52B shows the Positive and Negative Loop Comparisons BlockDiagram.

[0171]FIG. 52C shows the Noise Processor Block Diagram.

[0172]FIG. 53A shows an Example of Single Negative and Positive Loop.

[0173]FIG. 53B shows an Example of a Sum of Offsets Negative LoopFeedback.

[0174]FIG. 53C shows an Example of a Product Chain Negative LoopFeedback.

[0175]FIG. 53D shows a Detailed Negative Loop Comparison and Δ to GainConverter.

[0176]FIG. 53E shows an example of Loop Balancing with Multiple DelayCompensation Elements.

[0177]FIG. 53F shows an example of Loop Balancing with a common DelayCompensation Element.

[0178]FIG. 54A shows a Preferred Embodiment Noise Processor forCompander Method.

[0179]FIG. 54B shows a Preferred Embodiment Noise Processor for VolumeControl Only Method.

[0180]FIG. 54C shows an example of a Simple Noise Processor for VolumeControl Only Method.

[0181]FIG. 54D shows an example of a Multiple Positive Loop Input NoiseProcessor for Multiple Independent Companders.

[0182]FIG. 54E shows an example of a Three Positive Loop Input NoiseProcessor.

[0183]FIG. 54F shows an example of a Single Positive Loop Input NoiseProcessor for Multiple Independent Companders.

[0184]FIG. 54G shows an Example of Negative Loop Error Correction.

[0185]FIG. 54H shows a Noise vs. Microphone Graph.

[0186]FIG. 55A shows a Noise Compensation Variable Attack/Release LinearResponse.

[0187]FIG. 55B shows a Noise Compensation Variable Attack/Release LogResponse (Ear Response).

[0188]FIG. 55C shows a Noise Compensator Attack/Release Processor BlockDiagram

[0189]FIG. 56 shows an exemplary Embodiment Attack/Release Module forNoise Processor—Part A.

[0190]FIG. 57A shows an exemplary Embodiment Attack/Release Module forNoise Processor—Part B.

[0191]FIG. 57B shows Tracking Adjusting Noise Filter Signals.

[0192]FIG. 58 shows the Top Level Noise Detector flow diagram.

[0193]FIG. 59A shows the Environmental Sensor Processing flow diagram,Part 1.

[0194]FIG. 59B shows the Environmental Sensor Processing flow diagram,Part 2.

[0195]FIG. 59C shows the Reference Signal Processing flow diagram, Part1.

[0196]FIG. 59D shows the Reference Signal Processing flow diagram, Part2. FIG. 60A shows the Loop Comparisons flow diagram.

[0197]FIG. 60B shows the Negative Loop Comparison flow diagram.

[0198]FIG. 60C shows a Positive Loop Comparison flow diagram.

[0199]FIG. 61A shows a Noise Processor flow diagram.

[0200]FIG. 61B shows a Corrections flow diagram.

[0201]FIG. 61C shows a Volume Control Offset Processing flow diagram.

[0202]FIG. 61D shows a Variable Attack/Release flow diagram.

[0203]FIG. 61E shows a Sensitivity Control flow diagram.

[0204]FIG. 62 shows an exemplary Embodiment of Coarse and Fine AcousticLoop Balance Processor—Two Stage Balancing Method.

[0205]FIG. 63 shows a Loop Balance flow diagram.

[0206]FIG. 64 shows a Set Minimum and Maximum Method: Step 1—SetMaximum.

[0207]FIG. 65 shows a Set Minimum and Maximum Method: Step 2—SetMinimum.

[0208]FIG. 66 shows a Set Minimum and Maximum Method: Step 3—Post SetVolume Change.

[0209]FIG. 67 shows a Set Minimum and Typical Method: Set Typical byAdjusting Volume Control.

[0210]FIG. 68 shows a Set Minimum and Typical Method: Alternative SetTypical by Adjusting Amplifier Gain.

[0211]FIG. 69 shows an exemplary automatic method of default settings.

[0212]FIG. 70 shows an exemplary automatic method for increasing maximumvolume.

[0213]FIG. 71 shows an exemplary automatic method for decreasing maximumvolume.

[0214]FIG. 72 shows an exemplary automatic method for increasing minimumvolume.

[0215]FIG. 73 shows an exemplary automatic method for decreasing minimumvolume.

[0216]FIG. 74 shows an alternative automatic method for settingdefaults.

[0217]FIG. 75 shows an exemplary automatic method for increasing minimumvolume in response to noise.

[0218]FIG. 76 shows Noise Level Compensation for Non-Compander Systems.

[0219]FIG. 77 shows a Conceptual Setup Command flow diagram.

[0220]FIG. 78 shows a Conceptual Setup Sensitivity Command flow diagram.

[0221]FIG. 79 shows a Conceptual Intelligent Volume Control Command flowdiagram.

DETAILED DESCRIPTION OF THE INVENTION

[0222] Referring next to FIGS. 3A and 3B, numerous aspects of theoverall system and process flow of the present invention can beappreciated at least generally. With particular reference to FIG. 3A,the system 300 can be seen, in broadest form, to comprise a plurality ofuser interface and signal processor configurator blocks 305A-n combinedwhich receives user control signals 310 as inputs, includingbidirectional remote links 310A, and communicates bidirectionally withsignal processor blocks 315A-o. Aside from the signals communicated bythe user interface block 305 to the signal processor block 315, the userinterface block 305 also provides user display signals 320. The signalprocessor blocks 315A-o receive additional signal inputs 325, which maycome from any of a variety of audio or similar sources, and generatedone or more signal outputs 330 which have been expanded, compressed, orotherwise modified in accordance with features described in greaterdetail hereinafter.

[0223] The system 300 operates continuously to provide dynamic controland adjustment of the audio outputs in accordance with both user inputsand ambient or environmental conditions, with the net result of anenjoyable audio experience where the audio signals are automatically anddynamically adjusted to conform to the user's wishes in variousconditions. Depending on user settings, for example, the present systemmay permit an audio signal to be reduced automatically in the presenceof a conversation so that the conversation continues unimpeded as thoseconversing enter a room. Alternatively, the audio system and process ofthe present invention may be set to increase volume up to apredetermined limit if an intrusive noise—a gardener's blower, forexample—suddenly intrudes on the user's audio environment.

[0224] In an exemplary embodiment of the present invention, the userinterface block 305 will include a microprocessor and—depending on theparticular microprocessor selected—may also include various A/D and D/Aconverters, buffers, drivers and related logic. Depending on theimplementation selected, the user interface block may also be arrangedto provide one microprocessor per listening environment, for example oneuser interface per room. As part of the user interface function, themicroprocessor may also be configured to perform other relatedfunctions, including configuration, resource allocation, and control.These functions will be discussed in greater detail hereinafter. Atypical microprocessor may be, for example, a Motorola 6805 or Intel8048.

[0225] The signal processor block 315, on the other hand, typically willinclude one or more digital signal processors (abbreviated hereinafteras “DSP”). The DSPs may be assigned in many different configurations,including for example either a multi channel/band serial pipelinedconfiguration or a parallel configuration with one DSP per channel orband. A typical DSP may be, for example Motorola 56300 or TexasInstruments TMS320C54X series architecture. Alternatively, and perhapspreferable in some applications, the signal processor block may beimplemented in analog circuitry, particularly where cost issues prohibitthe use of even an inexpensive DSP.

[0226] In the simplest case, both user interface 305 and signalprocessor 315 may be implemented in the same microprocessor or DSP oranalog circuitry.

[0227] Keeping in mind the broad system description of FIG. 3A, theprocess may be broadly appreciated from FIG. 3B. In particular, theprocess begins at step 350 with a conventional power-on reset. Theprocess advances to step 355 where the system is initialized inaccordance with previous defaults or user settings. In a typicalembodiment, noise compensation is initially disabled at step 355 aswell. Noise compensation is initially disabled to avoid unpredictablebehavior, since the state of the environmental noise cannot beaccurately determined until after power-on initialization and systembalancing. Once the system has balanced, as discussed hereinafter, noisecompensation is typically employed. In addition, noise compensation maynot be utilized in some implementations.

[0228] The process of FIG. 3B thereafter advances to the user interfacefunctions of step 360, described in greater detail hereinafter. Ingeneral, the user interface step allows user-influenced operatingconditions. The process then advances to process the input signalpre-processing functions at step 365. The input signal pre-processingfunctions, described in greater detail hereinafter at FIGS. 8A through19, vary with implementation but generally include a plurality offunctions which are best handled prior to the compander functions. Suchfunctions may include all or only some of the following: input signalprocessor, environmental sensor adjustments and noise extraction,statistical engine, transform engine, and central power estimator mixer.In general, the multiple iterations of each function or combination offunctions may be used to provide multi-band or multichannel processing;alternatively, the process can be executed in parallel rather than bymultiple iterations.

[0229] Following the processing of the input signal pre-processingfunctions, the invention advances to step 370 for processing of thecompander functions. The compander functions may, depending on theapplication involved and the acceptable cost, include some or all of:bandsplit filtering and scaling, half-wave signal processing,synchronization, setting gain, and, in some implementations, providing asoft clip function. The features and functions are described in greaterdetail hereinafter in connection with FIGS. 20A through 40C. As with theprior functions, multiple iterations may be required for multiband ormultichannel implementations; alternatively, a parallel implementationmay be used.

[0230] Following completion of the compander processing at step 370, theprocess advances to the volume control and pre-mixer functions at step373. The volume control and pre-mixer functions may involve, dependingon the implementation and the acceptable cost, various aspects of signalmixing and volume (i.e. signal amplitude) control, including some or allof multi-input signal mixing and scaling, volume control, andcalibration signal selection.

[0231] Following completion of the volume control processing at step373, the process advances to the output processing functions at step375. The output processing functions may involve, depending on theimplementation and the acceptable cost, various aspects of channelprocessing and band processing, including bandsplit filtering andscaling, signal combining, soft clip, amplification, output conversion,reference generation, and other similar functions. As before, theprocess may be iterative depending on the number of channels and whetherone or more channels have been split into various frequency bands.

[0232] Once the steps 360 through 375 have been completed once, theprocess loops back to step 360 to process the next signals or, fordigital signals, the next input samples.

[0233] While FIG. 3B shows a typical process flow, it can be appreciatedthat other alternative process flows may be realized. For example, notall process steps may be required, the process order may be mixed,process steps may be executed in parallel, and multiple occurrences ofprocess steps may be used, for example, to implement a multilevelcompander and volume control chain.

[0234] Referring next to FIG. 4, the overall system architecture of ageneralized embodiment of the present invention may be betterappreciated. In general, the architecture comprises a plurality offunctional blocks connected by a system bus 400, which, for purposes ofexplanation may be thought of as comprising a control bus portion 400Aand a signal bus portion 400B. However, the system bus 400 may beconfigured simply as a single bus over which both control packets andsignal packets pass. A typical such implementation may be an IEEE 1394network or other suitable networking configuration. Note that System Bus400 can change in its implementations throughout the distributed system.

[0235] Regardless of how the system bus 400 is implemented, thefunctions of the system bus with respect to the functional blocks ofFIG. 4A, including its control functions and signal functions, may beappreciated from Table A, below. The particular functional blocks willbe described in greater detail hereinafter. TABLE A Control Bus - UserInterface Control Signals Module Inputs Outputs User Statistical EngineResource Allocation Signals 630 Interface 405 Flags & Data 1320 # ofChannels 640 (FIG. 6A) Current System # of Bands 640 Allocation 630 AddComputational Modules 630 External Device Remove Computational ModulesStatus 635 630 Reconfigure System 630 Automatic Reconfiguration UsingStatistical Engine Flags & Data, 640 −> Reset Statistical Engine Flags650 External Configuration Radio/TV station selection 635 DVD/CD trackand disk selection 635 MP3 song selection(s) 635 Transform Engine 410Select Input Dynamic Range 645 User Set Minimum Output Level 645 UserSet Maximum Output Level 645 User Volume Control 645 Statistical Engine415 Reset all or specific histograms/ analyzers 650 User StatisticalEngine Internal Configuration Signals Interface 405 Flags & DataCompander Module: (FIG. 6A) 1320 & 1325 Synchronizer 2045 (FIG. 21A)Input Clip Indicator Initialize Wave Buffer and Clip Counter InitializeGain Buffer 990 Compander Module: Variable Clip Event Attack/Release2275 (FIG. 24A) Counter and Internal Configuration 640 (e.g. Indication4020 Compander Operating Parameters- Slope 2290, User Selects, SelectionTable Data), Attack/Release Parameters 2274 (Variable A/R SegmentKneepoints, Load Var. A/R Parameter) Compander Module: Soft Clip 2035Clip Threshold, Reset Clip Counter Compander Module: Local PowerEstimator 2280 (FIG. 25A, B) Local Post Power Estimator Parameters andCoefficients, Algorithm Select Compander Module: Linear to GainTransform 2285 (FIG. 26A) Compander Gain Calculate Parameters 2290(Segment Boundaries/Kneepoints, Linear Parameters, e.g. Slope M and BOffset, Non-Linear Parameters, User Select Preferences) User StatisticalEngine Internal Configuration Signals Interface 405 Flags & Data 1320Noise Compensation 465 (FIG. 51) (FIG. 6A) Loop Closure Input Adjustblocks 5300 gain Done 6260 values (FIG. 52A) Input Clip Indicator NoiseSensitivity Control 5440 and Clip Counter Start Loop Closure 6255 990User Select Preferences for Clip Event Counter Variable and Indication4020 Attack/Release (shorter/longer responses) Transform Engine DataVolume Control/Pre- Mixer 445 (FIG. 41B): Volume Control Pre-MixerLevels 4210 Calibrate 640 Volume Control Setting 1640 Output SignalProcessor 475 (FIG. 43A): Input Mixing/Summing Control Analog/DigitalSelection Data Word Scaling Select Data packet lengthCalibrator/Annunciator 420 (FIG. 50): Internal Configuration 640 (e.g.Calibrate, Calibrate Signal Select, Annunciator Command, AnnunciatorMessage Select) Central Power Estimator 455 (FIG. 18): Parameters andCoefficients Algorithm Select Input Signal Pre-Processing & Input LevelAdjust 440 (FIG. 8A): Reset, 0 dB Level, Load Gain Value, Save GainValue, Input Select, Filter Parameters & Equalization Constants SoftClip 2035: Set Clip Threshold, Reset Clip Counter Compander ControlSignals The adaptive variable dynamic compander has multiple moduleswith signals connecting to the system bus 400. Control Bus - CompanderControl Signals Compander Module Inputs Outputs Synchronizer InitializeWave 2045 Buffer (FIG. 21A) Initialize Gain Buffer Variable InternalAttack/Release Configuration 640 2275 (e.g. Compander (FIG. 24A)Operating Parameters- Slope 2290, User Selects, Selection Table Data),Attack/Release Parameters 2274 (Variable A/R Segment Kneepoints, LoadVar. A/R Parameter) Soft Clip 2035 Clip Threshold Clip Event Counter andIndication Reset Clip Counter 4020 Local Power Local Post Power Est.2280 Estimator (FIG. 25) Parameters and coefficients Algorithm SelectGain Calculate Input Power 2620 Final Gain 2050 (FIG. 26A) (e.g. FinalPower Log Input Power & Selected Estimators 2283, Segment 2287 HalfCycle Peak Value 2289, Intermediate Power Estimate 2279), Compander GainCalculate Parameters 2290 (Segment Boundaries/ Kneepoints, LinearParameters, e.g. Slope M and B Offset, Non-Linear Parameters, UserSelect Preferences) Miscellaneous Module Control Signals Control Bus -Control Signals Module Inputs Outputs Noise Channel Reference CompanderNoise Floor 5110 Compensator Out 4310 Volume Control Noise Offset 5115465 (FIG. 51, Noise Sensitivity Loop Closure Done 6260 52A-C) Control5440 Start Loop Closure 6255 Input Adjust blocks 5300 gain valuesInternal Configuration 640 Transform Internal Compander Gain CalculationEngine 410 Configuration 640 Coefficients 2290 (FIG. 15) (e.g. setmaximum Volume Control Setting 1640 level, set minimum Vol. Cntl.Pre-Mixer Levels 4210 level, user volume Attack/Release Parameters 2274control, input dynamic range) Compander Noise Floor 5110 Volume ControlNoise Offset 5115 Statistical Engine Flags & Data 1320 StatisticalStatistical Engine Statistical Engine Controls 650 Engine 415 Controls650 (e.g. (e.g. Reset Input Level Adjuster, (FIG. 13) Reset all orspecific Input Dynamic Range) histogram/analyzers) Statistical EngineFlags 1320 Inputs (e.g. (e.g. New Kneepoints, Active Power Estimates)Channels, Inactive Channels) Statistical Data 1325 Volume Volume ControlControl/ Pre-Mixer Levels Pre-Mixer 445 4210, Calibrate 640, (FIG. 41B)Volume Control Setting 1640 Output Signal Input Mixing/ ChannelReference Out 4310 Processor 475 Summing Control (FIG. 43A)Analog/Digital Selection Data Word Scaling Select Data packet lengthCalibrator/ Internal Annunciator Configuration 640 420 (FIG. 50) (e.g.Calibrate, Calibrate Signal Select, Annunciator Command, AnnunciatorMessage Select) Central Power Inputs 1800 (Exp. Global Power Estimates2281 Estimator 455 Local Power (FIG. 18) Estimates 2282 and Global PowerEstimates 2281) Parameters and Coefficients Algorithm Select InputSignal Reset, 0 dB Level, Exceeded Adjustment Range PreProcessing LoadGain Value, Input Clip Indicator and Clip and Input Level Save GainValue, Counter 990 Adjust 440 Input Select, Filter (FIG. 8A) Parameters& Equalization Constants Soft Clip 2035 Set Clip Threshold Clip CounterValue Reset Clip Counter Power Estimator Signals Control Bus - PowerEstimator Signals Local Power Local Power Exported Local Estimator 2280Estimators 2279 Power Estimates 2282 (FIG. 25) and/or Initial PowerFinal Power Estimates 2283 Estimate 2273 Global Power Estimates 2281Central Power Local Exported Global Power Estimates 2281 Estimator 455Power Estimators (FIG. 18) 2282 Global Power Estimates 2281 MonitorSignals Control Bus - Monitor Signals Compander Attack/Release HalfCycle Power Estimates 2278 Modules 450 Parameters 2274 Initial PowerEstimates 2273 (FIG. 22) Global Power Local Intermediate Estimates 2281Power Estimators 2279 Gain Calculate Exported Local Parameters 2290Power Estimate 2282 Final Power Estimator 2283 Log Input Power, SelectedSegment 2287 Final Gain 2050

[0236] Monitor signals are any not already covered under the UserInterface, Compander, Miscellaneous Module, and Power Estimator SystemBus signals that may be monitored by the Statistics Engine or UserInterface.

[0237] As noted above, the foregoing aspects of the system bus 400 willbe described in greater detail hereinafter in connection with themodules associated therewith.

[0238] With continued reference to FIG. 4, the user controls 310,including the bidirectional remote links 310A from FIG. 1, can be seento be applied to a user interface stage shown generally at 405. Usercontrols may include but not limited to keyboards, keypads, touchscreens, and infrared and radio frequency remote controls. Remote links310A typically connect to another computer or control system, such as anInternet website or customer service computer, typically through the useof an analog or digital modem or other serial or parallel interface, forpurposes of running remote diagnostics, obtaining software updates,obtaining music or other sound data, and so forth. Depending on theparticular implementation, the user interface stage 405 may beimplemented as one user interface module per room or other suitablepartition, in which case there will be a plurality of user interfacemodules 405A-t. The number of user interface modules 405A-t can varywith the number of listening environments, and is typically configuredas one per listening environment although a smaller number is alsoacceptable in at least some embodiments. In addition to receiving usercontrol signals 310, the user interface stage 405 receives varioussignals from the remainder of the system via the system bus, as shownabove in Table A. The signals received by the user interface stage 405include a statistics engine flags and data, transform engine controls,resource allocation signals, monitor signals, noise floor levels, andconfiguration data. Ultimately, the user interface stage generatessignals to drive the various displays 320, which may be of any suitabletype. For example, the displays may include LED's, Braille generators,annunciators, remote link outputs, or any other suitable signalindicator for indicating system operation. In addition, the userinterface stage 405 provides various system bus signals as shown inTable A.

[0239] The user interface 405 also allows allocation of various systemresources to optimize the performance of higher-end implementations ofthe system including assignment of new and existing system resources.For example, in a home environment involving several rooms and whereinthe system of the present invention includes a plurality of processors,which can be timeshared, and related resources, it may be desirable todedicate multiple modules to a primary listening room, and allocate onlylimited resources to other areas. That requirement may later change, inwhich case it may be desirable to reconfigure the system to reallocatemultiple modules to different or multiple listening areas. The userinterface 405 permits such defining and assignment of system resources,and may be implemented by appropriate signals on the system bus 400, acrossbar switch, a routing table or network, or other suitable means.

[0240] Also connected to the system bus 400 is a transform engine stage410. The transform engine stage 410 operates to establish an operatingconfiguration for the system overall, including responding to the usercontrols which are provided through the user interface stage 405. Thetransform engine typically accepts conventional inputs such as volume,noise level and minimum volume levels and converts them to operatingparameters such as linear or non-linear multiplier values, volumecontrol values, compander operating parameters, gain calculation values,and so on. The transform engine stage 410 may be comprised of aplurality of transform engine modules 410A-u, where some or all of thelistening environments within an overall system may have a transformengine module 410 associated therewtih.

[0241] In at least some implementations, the user interface stage 405and transform engine 410 provide the user the opportunity to establishperformance parameters for the remainder of the system, as will bediscussed in greater detail hereinafter. In particular, the user can setcurrent volume, can set minimum and maximum volumes for audio signals,can determine whether the system outputs should dominate environmentalnoise (noise compensation) or should diminish in the presence of ambientsounds such as conversation (signal muting), can determine how quicklythe system responds to changes in ambient conditions, can set dynamicrange, and so on. In some implementations, however, it may be preferableto establish system defaults, in which case the user may not need toprovide any controls except possibly volume. In other implementations,particularly lower end implementations involving single channels orsimplified processing, it may be desired not to provide any usercontrols including configuration controls.

[0242] A statistical engine stage 415 may also be provided to trackvarious historical operating parameters. The statistical engine stage,which need not be implemented in all embodiments, may be implemented inone or more modules 415A-s where each statistical engine can theninstruct the user interface to change performance parameters for theremainder of the system by the use of generated flags. In addition, theuser interface can use data supplied by statistical engines tointelligently change performance parameters for the remainder of thesystem.

[0243] Further, a calibrator/annunciator stage 420, also implemented asone or more calibrator modules 420A-c, may also be provided to permitsetting of minimum and maximum volume, calibrating environmentalcompensation, and to balance multiband or multichannel systems. Anannunciator function may also be implemented in the stage 420 to provideaudible instructions or other comments to the user. In some embodiments,the calibrator function is of particular importance since it is helpfulfor balancing the system to provide effective noise compensation or forsetting minimum and maximum volume levels. At calibration, thecalibrator stage 420 generates a white noise or other appropriate signalwhich is permitted to override other system input signals so it alone issupplied as the system outputs. The resulting output is then measuredand the system set accordingly. The annunciator function—which permitsaudible instructions or comments by the system to the user—may beimplemented by bypassing portions of the signal path, although thecompander will typically still be used in at least some embodiments suchas, for example, applications involving the hearing impaired. Thecalibrator/annunciator functions may not be required in all systems. Thecalibrator/annunciator stage 420 communicates bidirectionally with theremainder of the system via the system bus 400, as described in Table A.

[0244] Having discussed the general operation of the user interfacestage 405, transform engine 410, statistical engine 415 andcalibrator/annunciator stage 420, the second major portion of the systemof the present invention is the signal path. Still with reference toFIG. 4, digital signal inputs 430 or analog signal inputs 435 areprovided to an input signal pre-processing stage 440. The digital signalinputs 430 may comprise one or more sources 430A-w. As with the userinterface stage 405, the input signal pre-processing stage 440 may beconfigured as one or more pre-processing modules 440A-i depending on thedesired implementation. In a robust implementation, the input signalpre-processing stage 440 may be configured as one module per channel,for example, although other signal partitions will be readily apparentto those skilled in the art. The general function of the input signalpre-processing stage 440 will be discussed in greater detail hereinafterin connection with FIGS. 8A-8F, 9A-C and 10A-10D, but is basically toallow multiple input signals to be multiplexed, to convert analog inputsignals to digital form, to synchronize relevant events and data, and soon.

[0245] The input level adjust stage also serves to establish what aregenerally referred to as “0 dB levels,” as well as to load and to saveinput gain values. For purposes of the present description of anexemplary embodiment, 0 dB levels are set to permit optimal companderoperation, and in a preferred embodiment are set so that the largestamplitude (or peak) input signal available from a given source maps tothe maximum acceptable amplitude digital signal for the system. Thisconcept is discussed in greater detail in connection with FIG. 8B,hereinafter. The maximum acceptable digital signal may be, for example,the maximum non-distorting signal permissible in the system for therange of frequencies of the system, or may be set at a different level,for example somewhat less than the maximum to allow a certain amount ofheadroom, to permit management of output signals which would otherwisebe distorted through clipping. Thus, a peak analog signal from a firstinput may, for example, be two volts peak-to-peak and is mapped by theinput level adjust stage to the maximum amplitude digital signalpermissible by the system. But a second input source, for which the peaksignal may be only 0.5 volts peak-to-peak, is also mapped by the inputlevel adjust stage to that same maximum amplitude digital signal, orwhat may be thought of as the “compute space.” That maximum amplitudedigital signal is defined, for the exemplary embodiment describedherein, as the “0 dB level.” Thus, the volume of signals from varyinginput sources will all map to the same digital amplitude for furthersystem processing.

[0246] In addition, in some configurations, the input signalpre-processing stage 440 may include a band-splitting function to, forexample, divide the incoming signal into a plurality of frequency bandsfor subsequent processing, or to provide filtering or equalization.

[0247] Downstream of the input signal pre-processing stage 440, theinput signals are provided to a dynamic compander stage 450, which inmany embodiments will cooperate closely with a central power estimatormixer stage 455 to permit multiple bands or channels of signals withoutaltering relative spatial information or causing signal distortionbetween the various bands or channels. While the power estimator mixerstage 455 is shown separately from the compander stage 450, in someembodiments the power estimator mixer function may be incorporated intothe compander. In general, as the numbers of channels or bands increasesit becomes more efficient to utilize a centralized power estimatormixer. Thus, for the exemplary embodiment of FIG. 4, a centralizedversion of the power estimator mixer has been shown. In some instances,a centralized power estimator mixer stage will cooperate with multiplelocal power estimator mixers associated with each compander stage, asdiscussed in greater detail hereinafter.

[0248] Referring still to the compander stage 450 in general, thefunction of the compander stage is to match the input source dynamicrange to the listener's dynamic range by either expanding or compressingthe input signals as appropriate in accordance with the user or systemcontrol signals provided either by the user interface 405 or inaccordance with environmental levels provided by the noise extractorstage 465. The compander stage forms a central portion of manyembodiments of the overall system, and in a presently preferredembodiment operates with low distortion. One element for providingparticularly low distortion is the implementation of a variable attackand release function, which will be discussed hereinafter but basicallypermits the compander to dynamically adjust for changingparameters—either environmental or input signal changes—to maintainoutput signals within predetermined limits. As with several of the otherstages, the compander stage may comprise a plurality of compandermodules 450A-e, according to the number of channels or other signalpartitions utilized in the particular embodiment. In an exemplaryembodiment, the compander stage 450 will include a variable attack andrelease function to permit rapid adjustment of compander parameterswhile maintaining low distortion. In addition, as noted above, inmultichannel embodiments the compander may include a local powerestimator mixer.

[0249] The power estimator mixer stage 455 may be implemented as one ormore modules 455A-p where, for example, each module may be associatedwith a listening environment. The power estimator mixer stage operatesto permit multiple compander stages without altering relative spatialinformation or causing signal distortion between the various bands orchannels.

[0250] In an additional feature, the compander stage 450 responds toinputs from user interface stage 405, transform engine stage 410 andenvironmental inputs from a noise extractor stage 465 discussedhereinafter to implement the user's selection of either noisecompensation or signal muting. In noise compensation, the audio signalis automatically compressed when environmental or ambient noise occurs,thus ensuring that the low volume portions of the audio signal can beheard despite the ambient noise. In signal muting, the ambientnoise—which includes conversation or other high priority environmentalsounds—is given priority over the system's audio signal, thus allowingthe conversation to be heard even over the loudest audio signal portion.

[0251] The input signal pre-processing stage 440, compander stage 450,and power estimator mixer stage 455 each communicate bidirectionallywith the remainder of the system via the system bus 400.

[0252] A noise extractor stage 465 may also be provided in someembodiments. As with the prior stages, the noise extractor stage 465 maybe implemented as a plurality of modules 465A-m. The function of thenoise extractor stage is to provide an indication of the environmentalnoise level in the listening environment through environmental inputs,typically one or more microphones or other sensor inputs positioned inthat listening environment. The environmental inputs will typically havea combination of environmental noise and speaker output components. Bycomparing the environmental inputs with the estimated output power ofthe system—i.e., a representation of the speaker outputs, also referredto as the system reference signal—the environmental noise component canbe isolated and a signal indicative of that noise component can be fedback to the remainder of the system to adjust output levels accordingly.The system reference signal may be, in at least some embodiments, acombination of reference signals, e.g. one per channel to facilitateefficient acoustic loop balancing, typically on a speaker-by-speakerbasis.

[0253] Balancing, also referred to as acoustic loop balancing, systembalancing or system calibration, is typically performed once for a givensystem configuration and acoustic environment—that is, a balancingtypically is performed only when something about the systemconfiguration or the listening environment changes. Balancing sets theenvironmental input to be substantially equal to the system referencesignal in the absence of environmental noise. Once balanced, the correctamount of environmental noise can be determined from the environmentalinput. To ensure that the reference signal represents the power emittedfrom the speaker, any post-balancing signal changes (e.g. tone controls)are typically done before the reference signal is generated so that suchchanges are included in the loop, allowing the system to remainbalanced.

[0254] It will be appreciated that the process of adjusting output inaccordance with the noise extractor stage 465 essentially forms apositive feedback loop. As will be discussed in greater detailhereinafter, a negative feedback loop is also formed, to compensate forchanges in the listening environment, for tolerances in theenvironmental sensors and associated components, and also to eliminateany “gain chase” issues. The noise extractor stage 465 communicates withthe remainder of the system via the system bus 400.

[0255] Subsequent to the compander stage and noise extractor stage, theaudio signals are provided to a volume control stage 445A-v and anoutput signal processor stage 475A-o via the signal bus 400B, withappropriate control signals as identified in Table A supplied by thecontrol bus 400A. The volume control stage 445 and output processorstage 475 operate mainly to convert the signals into sound atappropriate volumes and output levels, and may include both signalmixing and amplification, depending upon implementation. The sound maybe supplied to discrete sound outputs 480 or other analog devices 485;alternatively the output signal may be provided in digital form tovarious digital receivers 490A-x.

[0256] From FIG. 4, the overall structure of the present invention maybe appreciated. It can be appreciated that many different configurationsof the elements of FIG. 4 can be realized. For example, not all elementsmay be required, the element order may be mixed, elements may beexecuted in parallel, and multiple occurrences of elements may be used.FIGS. 5A-I give examples of a variety of configurations.

[0257] With reference to FIG. 5A, the application of the system to auser environment can be better appreciated. The partitioned audio systemof the present invention can be seen to include analog sources, orinputs, 435 provided to centralized partitioned signal processing 500,which also receives digital inputs 437 from digital sound producingdevices 430 and remote commands, data, or programs from remote link310A. The centralized partitioned signal processing 500 may include,depending upon the particular embodiment, one or more input signalpre-processing stages, volume control and pre-mixer stages, companderstages, central power estimator mixer stages, and so on as discussed inconnection with FIG. 4.

[0258] A significant advantage of the expandable audio server networkshown in FIG. 5A is its ability to minimize redundant equipment. Byproviding centralized partitioned signal processing 500, other roomlocations are able to share input and output devices, and computationalmodules. By using an audio/visual file server 520, once a given sequenceof sounds has been acquired, it can be stored on the server, and playedat any location connected to the network. Prior-art solutions oftenrequired having standalone complete sound systems in each room orrequired every device to be located at a single common location.

[0259] Rooms connected to the network may have a wide range ofcapabilities, ranging from the very simple stereo only configuration ofRoom C 505C, to the complex, three-band, stereo, noise compensation forRoom A 505A. Equipment may also be distributed and shared across thenetwork, such as the additional devices shown in Room B 505B.

[0260] The sound in each room can be optimized in a wide variety ofmanners to meet listener preferences. For example, a different listenerdynamic range can be specified for each room. Rooms that have noiseextractors 465 can be configured to automatically transform the soundbeing played when environmental noise interferes with listeningconditions. Prior-art solutions only provided a conventional volumecontrol.

[0261] The audio/visual file server 520 shown in FIG. 5B makes use ofthe input signal processor 440 and output signal processor 475 tointerface between the system bus 400 and the file server controller 525.While the file server controller 525 and digital mass storage 530 mightbe a dedicated file server, it also could be provided by means of a PC,network appliance, or any other digital device able to provide and/oraccept digital information used to encode sound. Often sound would bestored in a compressed format using MP3 or AC-3 algorithms, andsubsequently decoded by the file server controller or other means.

[0262] Advantages of the audio/visual file server 520 are that itminimizes redundant input devices, reduces the labor required to playmusic, eliminates needing to store recordings near sound reproductionequipment, and allows multiple, concurrent streams of audio to beprocessed by the centralized partitioned signal processing 500 and thensent to each room.

[0263]FIG. 5C shows that a centralized partitioned signal processing 500may include all of the modules articulated in FIG. 4, with the normalexception of the noise extractor 465 that typically must be located ineach room in order to measure the local ambient noise level. Multiplerooms can make concurrent use of the resources provided by centralizedpartitioned signal processing 500. Resources can be timeshared ordedicated as desired by its users. Prior-art solutions provide fixedallocations of resources, preventing their full use.

[0264] Typically a room with noise compensation would include theminimum modules shown in FIG. 5D. The discrete sound outputs 480 aredriven by the output signal processor 475 which receives signalinformation from the centralized partitioned signal processing 500. Thenoise extractor 465 detects the total environmental input 470 and sendenvironmental noise information to the centralized partitioned signalprocessor 500. The user interface 405 allows the user to control theoperation of the modules distributed throughout the network by means ofuser controls 310 in the room, as well as to make requests of thecentralized resources, such as to play a particular song from the fileserver. When using this room configuration, the centralized partitionedsignal processing 500 would provide all other sound processingcapabilities.

[0265] To implement Room B 505B, a room with local input/outputcapabilities, but without noise compensation could be implemented asshown in FIG. 5E. This figure is similar to FIG. 5D with additionalmodules to provide input/output capabilities, but lacks the noiseextractor 465. A digital sound producing 430 and sound accepting device490 facilitate local connection of devices such as a portable digitalrecorder. A recordable CD player might be input to the room by discreteanalog input signals 435, input signal processor 440, and output viaoutput signal processor 475, and analog accepting devices 485. Anuniversal serial bus (USB) device such as a MP3 player might beinterconnected to the digital sound producing device 430 and digitalsound accepting device 490. These modules allow the user to have localdevices while making use of the centralized partitioned signalprocessing 500 and file server 520 resources.

[0266]FIG. 5F indicates how a “smart speaker” might be implemented.Using the compander 450, volume control and pre-mixer 445, and outputsignal processor 475, a speaker can be implemented, that allows matchingthe dynamic range of the sound input with the output dynamic rangedesired by the listener, speaker equalization, and easy installation.The smart speaker generates a reference out 4310 signal to allow its usewith a noise extractor 465 to facilitate its use in systems performingautomatic noise compensation.

[0267] Typically the smart speaker will be connected to the otherportions of the system via a single cable, easy to install bus suitablefor substantial distances. A system bus translator 535 converts thelocal system bus 400 to signals or packets sent on interfaces such asIEEE 1394, Ethernet, or other means having adequate bandwidth.

[0268] The partitioned signal processing system can also be used toimplement a superior hearing aid as shown in FIG. 5G. This example alsoshows how multiple levels of companders can be used.

[0269] In a hearing aid, a microphone provides a discrete analog inputsignal 435. This signal is processed by the input signal preprocessing440 that processes and converts it into a 0-dB adjusted signal 860. Thissignal is provided to a compander that acts to limit its maximumamplitude without causing signal distortion, and generates a wide bandpower estimate 540 of the signal. As shown in FIG. 8A, the input signalpre-processor 440 contains a band-split filter that may be usedseparately. This band-split filter is now used to split the limitedsignal into three different frequency bands.

[0270] One compander 450A,B,C is used for each frequency band, allowingeach to be individually compressed or expanded by different amounts.Using the central power estimator mixer 455, the three outputs from thecompanders are adjusted with relation to each other and the previouslycalculated wide band power estimate signal. This allows more optimalcorrection of an individual's hearing loss. The calibrator 420 can beused to determine the optimal signal-processing configuration for agiven user's hearing loss.

[0271] The outputs of the variable dynamic companders 450 are providedto a volume control & pre-mixer 445 that combines the three bands into asingle composite signal. This signal in turn is provided to the outputsignal processor 475 that generates the signal supplied to the discretesound output 480, in this case a miniature speaker. User controls 310,such as volume, work with the volume control and mixer 445, transformengine 410, statistical engine 415, and compander 450, to provideoptimized signal intelligibility no matter what the volume setting.

[0272] In contrast, prior art hearing aids change the overall volume ofthe signal and interfere with the optimal correction. In addition, theadaptive dynamic compander 450 provides virtually distortion freecompanding, providing superior sound quality over prior art solutions.

[0273]FIG. 5H shows a noise compensating, dynamic range mapping stereoimplementation of the partitioned signal processing system that could beused in a radio or television. The input signal pre-processing 440 takesthe left and right discrete analog input signals 435, and produces two0-dB adjusted signals. Each signal is provided to a compander 450 thatcompresses or expands the signal as required. Local power estimatormixing in companders 450A and B provides automatic maintaining of thecorrect spatial balance between left and right channels.

[0274] The left and right signals from the compander are provided to twovolume control & pre-mixer 445A and B, where the overall volume level isadjusted. These two signals are sent to the output signal processor 475that produces the signals necessary for the discrete sound outputs 480,typically speakers, as well as the left and right channel referencesignals 4310A,B supplied to the noise extractor 465.

[0275] As the total environmental noise component of environmental input470 changes, the noise extractor generates a noise signal that is usedby the transform engine 410, to control the compander 450 and volumecontrol 445 to produce signals with a dynamic range appropriate for thecurrent acoustic environment.

[0276] The calibrator 420 is used in a method for calibrating theoperation of the system. The statistical engine 415 monitors the overalloperation of the system, and is periodically used by the transformengine 410 and/or user interface 405 to adjust parameters of the system.Similarly, the user interface 405 allows the user to adjust and modifyoperation of the system.

[0277] Prior art implementations only allow changing the maximum volume.They are unable to do dynamic range mapping or accommodate environmentalnoise.

[0278]FIG. 51 shows the use of multiple layers of volume controls andpre-mixers to implement an 8 channel studio mixer. The signalpre-processing 440 enables all of the discrete analog input signals 435to be adjusted into a consistent 0-dB level. The first two sets ofvolume controls and pre-mixers 445A-J enables any portion of the eightsignals to be combined into a left or right channel.

[0279] Two companders 450A and B allow the left and right channels to becompressed or expanded, and use local power estimator mixing to maintainthe spatial balance between the two channels. Companding allow adjustingthe output dynamic range to that of the recording or transmission media.A third set of two volume controls and pre-mixers 445K,L allow theoverall amplitude of the stereo channels to be changed. Last, an outputsignal processor 475 is used to produce the desired analog outputs 485.

[0280] Adjustment of the volume control & pre-mixer 445 levels andcompander 450 functions are provided by the transform engine 410A-k. Thecontrol of the transform engines is provided by the statistical engine415A-s and user interface 405.

[0281] This example of a mixer provides virtually distortion freecompanding with superior spatial balancing. The input level adjustinggreatly simplifies initialization and setup of the mixer versus priorart implementations.

[0282] The individual elements, and the processes associated with them,will now be discussed in turn.

[0283] Referring first to FIGS. 6A-6C, the steps implemented in the userinterface stage, and the associated logic, may be better understood. Inparticular, FIG. 6A depicts in functional block form the logiccomprising the user interface, while FIG. 6B provides a simplifiedversion of the process implemented by the user interface of FIG. 6A.FIG. 6C provides a more detailed version of the process implemented bythe user interface of FIG. 6A. Various user signals 310 are provided touser input processor 605, the output of which is provided to a commanddecoder 610. A remote device 615, typically another computer or controlsystem, such as an Internet website or customer service computerconnected by the use of an analog or digital modem or other serial orparallel interface, may also provide inputs via remote link 310A to theuser input processor 605. The command decoder 610 is typicallyimplemented as a function within the microprocessor that operates toprovide the user interface stage 405, but could be implemented as anindependent microprocessor. The command decoder 610 also receivesvarious signals from the user interface via the system bus 400,especially statistical analysis, noise floor, and monitor portionsthereof, which can determine dynamically operating conditions andrelated operating parameters. The command decoder 610 then providesoutputs to drive a user output processor 620 and downstream displaydevices 320 and remote link 310A, as well as resource allocation signals630, external configuration information 635, internal configurationinformation 640, transform engine controls 645, and statistical enginecontrols 650. Each of these output signals is then provided to therelevant portions via the system bus 400.

[0284] The associated operational steps can be appreciated from FIG. 6B,where the process begins with a typical process start at step 652,followed by retrieving the commands and flag data at step 655, whichwill typically comprise user commands and statistical engine flag data.The command decoder then establishes the responsive operating conditionsat step 660, including supplying outputs to the display processor, etc.,at step 665. The process then exits at step 670.

[0285] The user interface process may be further illustrated withreference to FIG. 6C. The process begins at step 652 as in FIG. 6B, andproceeds to step 655. Step 655 may be seen to include step 655A, where acall is made to the user I/O process to get the User Command, followedby step 655B which calls for getting the statistical engine flags andassociated data as well as other relevant data and configurationinformation from control bus 400A. The process then advances to step 660from FIG. 6B, but may be seen to comprise several command processingevents. First, from a high level command and statistical engine flagprocessor function 660A of the command decoder 610, a plurality ofsequences may be initiated which may be generally characterized aseither no command, an operation command, a set-up command, or aconfiguration command. If no command is detected by the high levelcommand processor 660A, the process advances to step 665 on path 660B.

[0286] However, if an operation type command or flag is detected by theprocessor 660A, it is passed to the operation command decoder 660C,which in turn indicates to operation command execute step 660D whichcommand to execute. At step 660D, that command is then executed.Similarly, if the processor 660A detects a set-up command or flag, it ispassed to the set-up command decoder 660E which in turn indicates whichset-up type command should be called and executed at step 660F.Likewise, if a configuration type command or flag is detected at step660A, it is passed to configuration command decoder 660G for executionat configuration command execute step 660H. After execution of thesecommands, the system advances to step 660I, where the new system anduser settings are saved. The process of FIG. 6C then advances to step665, where the user output processor step can be seen to comprise thesub-steps of getting the signal processors status at step 665A, andupdating the user display and sending output to remote link 310A at step665B. The process then exits at 670 as discussed with regard to FIG. 6B.

[0287] Referring next to FIG. 7, the process by which a variety of themodules shown in FIG. 4 are incorporated into the system of the presentinvention can be better appreciated. The process of FIG. 7 begins atstep 700, and advances to step 705, when an input signal processorfunction is implemented. The function and its associated logic aredescribed in connection with FIGS. 8A through 10D. The process thenadvances to the noise extractor step 710, as shown generally at module465 in FIG. 4. The noise extractor step 710 is more fully described inconnection with FIGS. 11,12 and 51-63. The process then advances to thestatistical engine function 715, which is more fully discussed inconnection with FIGS. 13 and 14. Thereafter, the process advances to thetransform engine function 720, more fully described in connection withFIGS. 15-17, after which the overall process advances to the centralpower estimator mixer function, shown at step 725. The central powerestimator mixer function and associated hardware are more fullydescribed in connection with FIGS. 18 and 19. The process shown in FIG.7 then ends at step 730.

[0288] Turning next to FIGS. 8A-8G, FIGS. 9A-9C and FIGS. 10A-10D, theinput signal pre-processing 440 function and associated logic, which maybe broadly thought of as an input level matching system, may be betterappreciated. FIG. 8A shows in schematic block diagram form the logic ofthe input signal pre-processing 440 stage, while FIG. 8B showsgraphically the adjusting of various input signals 807 to the 0 dBlevel. FIG. 8C shows in schematic block diagram form the input leveladjust 810 portion of FIG. 8A, which may be broadly thought of as levelmatching logic. FIG. 8D shows in block diagram form the bandsplit andequalization function 825. FIGS. 8E through 8G show more detailconcerning digital, analog and hybrid implementations of the signalprocessor and gain cell, while FIGS. 9A-9C show alternatives of the clipdetector and analyzer logic implemented in the input level adjust stage.FIG. 10A shows the process flow of the input pre-processor stage, whileFIGS. 10B-10D show details of the process of FIG. 10A.

[0289] Referring first to FIG. 8A, a plurality of analog inputs 435 canbe seen to be provided to “select analog input group” logic 800. Typicalinputs may be any conventional audio inputs, including CD's, minidisks,TV, camcorders, PCs, radio receivers, and other similar devices, whilean input “group” refers to the one or more channels provided by theselected input or inputs. The “select input group” logic 800, whichbasically performs a multiplexer function, receives a select inputsignal via the control bus 400A which forms a portion of the system bus400. The select input signal may be provided by user inputs or othersuitable controls. The selected analog signal groups are then providedto analog signal processing logic 805. The analog signal processinglogic 805 may comprise a plurality of logic blocks 805A-m, typicallyconfigured as one per input channel in more robust embodiments.Processed analog signals, which are typically converted to digital form,are provided from the processing logic 805 to input level adjust logic810, which may likewise configured as a plurality of logic modules810A-p.

[0290] Similarly, a plurality of digital inputs 430 may be provided todigital form of “select digital input group” logic 815, which is againessentially a mux. The logic 815 also receives a “select input” signalfrom the control bus 400A, which causes selected inputs to be suppliedto digital data processing logic 820 which may comprise a plurality ofmodules 820A-q. Typical digital inputs may include any form of digitalaudio input, but particularly include USB and IEEE 1394 devices,including minidisk players, CD and DVD players, PC's, digital VCRs,satellite receivers, HDTV, IR repeaters, printers, and other similardevices. The digital signal processing logic 820 provides databuffering, packet disassembly and signal processing functions, afterwhich the processed signals are provided to the input level adjust logic810.

[0291] One important function of the input level adjust logic 810 is toset the 0 dB level for each of the input signals for the various devicesdescribed above, as discussed generally in connection with FIG. 4 butshown graphically in FIG. 8B. As noted previously, 0 dB levels are setto permit optimal compander operation, and in a preferred embodiment areset so that the largest amplitude (or peak) input signal available froma given source maps to the maximum acceptable amplitude digital signalfor the system. Thus, a peak analog signal 807 from a first input isshown at the upper left of FIG. 8B. That input signal, which may, forexample, be two volts peak-to-peak, maps to the maximum amplitudedigital signal 860 permissible by the system, or what may be thought ofas the “compute space.” That maximum amplitude digital signal 860 isdefined, for the exemplary embodiment described herein, as the “0 dBlevel.” In a feature of the present invention, a peak input signal froma second input 807 is shown at the lower left of FIG. 8B, and may beonly 0.5 volts peak-to-peak but still maps to the maximum amplitudedigital signal as shown at 860. Thus, the amplitude of signals fromvarying input sources will all map to the same digital amplitude forfurther system processing. It is to be understood that the 0 dB level isin some respects arbitrary, and may be set, for example, to the maximumamplitude digital signal the system can generate without distortion ashappens with low cost delta-sigma analog to digital converter; or may beset to a lower level which allows a certain amount of headroom beyondthe 0 dB level such as might be desirable for managing signals whichmight otherwise clip. This situation might occur when the input leveladjuster reaches it's minimum gain limit.

[0292] As noted above, the input level adjust logic is described ingreater detail below in connection with FIG. 8C; it also receivescontrol signals from the control bus 400A. As with logic 805, the logic820 may comprise multiple processors, for example one per input channel.Likewise, the input level adjust logic may also be configured asmultiple units, typically one per group of inputs or shared amonggroups. The output of the input level adjust logic 810, which typicallycomprises a group of 0 dB adjusted signals, is provided either directlyto the signal bus portion 400B of the system bus 400, or is provided tobandsplit and equalization logic 825A-x. The bandsplit/equalizationlogic is described in greater detail in connection with FIG. 8D, below.

[0293] Referring next to FIG. 8C, the input level adjust logic 810 canbe appreciated in greater detail. A plurality of processed analog ordigital signal inputs 1 to n, shown as 807A-n or 823A-n, which comprisewhat has been referred to previously as an input group, are provided tocorresponding signal processing and gain cells 855A-n, typically eithershared or arranged one gain cell 855 per group of inputs as discussedabove. The gain cells each provide an output signal 860A-n, typically a0 dB adjusted signal as previously discussed, an input clip signal900A-n output to clip detector analyzer 875 and also provide a clipsignal 877A-n, which is either input clip signal 900 or output clipsignal 905, to clip detector logic 865A-n, arranged one clip detectorper gain cell in a typical embodiment. As will be discussed hereinafter,input clip signals 900A-n are only required for direct input gaincalculations by clip detector analyzer 875. The outputs of all of theclip detectors 865A-n are typically provided to clip detector analyzerlogic 875, which receives a control signal to either reset or to load ina minimum gain value 870 for the selected input group (again, in apreferred embodiment, either the maximum or the last prior setting forthat input group) from the control bus 400A. The control bus 400A alsoprovides a path for storage of the latest minimum gain value 870 for theselected input group which can be stored locally or in some commonsystem memory. Clip detector analyzer 875 may also provide to controlbus 400A an input clip indicator signal typically for use by statisticalengines and user interfaces and may also contain a clip counter toprovide a clip count value. The clip counter may be reset via controlbus 400A. The clip detector analyzer logic 875 in turn provide a minimumgain value signal 870 to each associated signal processing and gaincell. In an exemplary embodiment, the same minimum gain value signal 870is supplied to each of the gain cells 855A-n to ensure equal gainsettings across all channels in the input group. The signal processingand gain cells 855, the clip detectors 865 and the clip detectoranalyzer 875 will all be discussed in greater detail in connection withFIGS. 8E-8G, 9A-9B, and 9C.

[0294] Referring next to FIG. 8D, the bandsplit and equalization stage825 of FIG. 8A can be better understood. In particular, the bandsplitand equalization stage 825 can be seen from FIG. 8D to comprise abandsplit filter 880, which receives the input signal from the signalbus 400B. The bandsplit filter divides the incoming signal into as manyfrequency or other bands as desired for the particular embodiment,resulting in n Band output signals from the filter 880, where n can varyfrom one to any higher integer. The n Band signals are provided to ascaling processor 885, which receives a equivalent number, from one ton, equalization signals from the control bus 400A. The scaling processor885, typically a plurality of multipliers, provides n Signal Out Bandoutput signals of equal or varying maximum amplitudes, which areprovided to the next stage via the signal bus 400B. Referring back toFIG. 8A momentarily, it will be appreciated that each of the elementsdescribed in FIG. 8A have now been described.

[0295] However, some additional details of FIG. 8C remain to be furtherdescribed. Referring now to FIGS. 8E, 8F and 8G, digital, hybrid andanalog signal processor and gain cells (855 in FIG. 8C) can be betterappreciated. FIG. 8E shows an exemplary digital cell, in which thesignal input 823 is supplied to a register 890, which also receives asampling clock 2105. The output of the register 890 is provided to ascaling processor 895 and provides an input clip signal 900. The scalingprocessor 895 also receives as an input a minimum gain value signal 870,and provides as its output an output clip signal 905, which also servesas an input to a register 910. The register 910 receives via bus 400Athe same sampling clock signal as the register 890. The gain cell ofFIG. 8E thus receives a digital input signal which is sampled by theregister 890, scaled by processor 895 in accordance with the signal 870,and then placed in register 910. The output of the register 910 servesas an output signal, typically a 0 dB adjusted signal 860.

[0296]FIG. 8F illustrates an exemplary form of hybrid signal processorand gain cell 855. The hybrid cell basically provides A/D conversion,followed by any desired signal processing, to yield the same outputs asthe cell shown in FIG. 8E. More specifically, the analog input 807signal is supplied to a digitally controlled amplifier (DCA) 920 and anAND converter 925. The DCA 920 also receives the minimum gain signal 870as an input. The output of the DCA 920 is provided to a second AIDconverter 930, which along with the AND converter 925 receives a convertclock signal 935. The convert clock can operate at the same rate as thesample clock 2105 on the bus 400A, but may also be configured to operateat a much higher rate, to permit multiple iterations before a 0 dBadjusted signal is supplied by register 950 for subsequent processing.This permits improvement in the accuracy of the register sample. Theoutput of the AND converter 925 is provided to optional signalprocessing logic 940, which in turn provides as its output an input clipsignal 900. The output of the AID converter 930 is provided as outputclip signal 905, which also provides an input to an optional signalprocessing stage 945. The output of the signal processing stage 945provides an input to a register 950, which also receives an input fromthe sample clock 2105 signal on bus 400A. The register 950 typicallyprovides as its output the 0 dB adjusted signal 860.

[0297] In FIG. 8G, an exemplary analog version of the signal processorand gain cell 855 can be better appreciated. An analog signal 807 inputis provided via signal bus 400B, but also is provided as the input clipsignal 900. The analog signal input is provided to a variable gainamplifier (VGA) 950. The VGA 950 also receives the minimum gain signal870 as a control input, and provides as its output both the 0 dBadjusted signal 860 and the output clip signal 905.

[0298] Referring next to FIGS. 9A-9B, the clip detector 865 of FIG. 8Ccan be better appreciated. The clip detector 865 can be implemented aseither an input clip detector or an output clip detector. Referringfirst to FIG. 9A, in which an exemplary version of an input clip circuitis depicted, a 0 dB threshold signal is provided as an input to adivider circuit 955, which generates a input threshold signal as itsoutput. The minimum gain value 870 is provided as a divisor input to thedivider 955. The threshold signal provides a negative input to acomparator 960, which receives its positive input from the input clipsignal 900. The comparator 960 generates a clipping signal indicative,in this exemplary embodiment, of whether the input clip signal isgreater than the threshold signal. Similarly, and referring to FIG. 9B,an exemplary version of the output clip circuit can be betterunderstood. The output clip signal 905 is provided as the positive inputto a comparator 970, with the 0 dB threshold signal providing thenegative input. The output of the comparator 970 serves as a “clip true”output signal, indicating whether the input clip signal has greatermagnitude than the threshold signal. It will be appreciated that theoutput clip detector shown in FIG. 9B does not require the divide logicused in the input clip detector of FIG. 9A, and for that the output clipdetector implementation offers some advantage over the input clipdetector implementation.

[0299] Referring next to FIG. 9C, the clip detector analyzer logic 875of FIG. 8C may be better appreciated. The basic function of the clipdetector analyzer logic is to determine whether gain should be reducedto avoid clipping and if true, to calculate a new minimum gain value.The “clip true” signals from each of the clip circuits 865 serve asinputs to an OR gate 975, which provides an output indicating whetherany of the “clip true” inputs was set. The input clip signals 1-n areprovided as inputs to “select maximum input amplitude” logic 980. Theoutput of the selection circuit 980 indicates the amplitude of themaximum input clip signal, which is provided as one input to a minimumgain value calculate block 985. The output of the “clip true” OR gate975 provides another input to the gain value calculate circuit 985 and,more specifically, signals the minimum gain value calculate logic 985that the minimum gain value 870 must be reduced, as well as enabling anew gain value calculation. The “clip true” OR gate 975 output is alsoprovided to input clip indicator and clip counter 990 to provide tocontrol bus 400A an input clip indicator signal and a clip count value.The clip counter may be reset via control bus 400A. The minimum gainvalue calculate logic 985 uses the maximum input clip signal from thelogic 980 to establish the new gain value by direct calculation,typically the result of dividing the 0 dB maximum amplitude value by themaximum input clip signal value. The logic 980 and the associated inputto the block 985 can be eliminated in at least some embodiments bysetting the gain reduction at some specified amount, for example afraction of the prior gain. This fractional reduction process is easierto implement (e.g., no A/D converter 925, FIG. 8F) and particularly wellsuited to iterative adjustment of the gain value 870, but offers lessimmediate precision than an implementation using the select maximuminput amplitude logic 980.

[0300] The control bus 400A also provides a communications path for theminimum gain value calculate logic 985, including a reset signal,setting a minimum gain value, and providing a new minimum gain value tobe shared with the remaining stages. As before, the new minimum gainvalue may be stored in memory local to the stage, or may be provided toa common shared memory accessible to each stage that requires the data.The output of the calculate circuit 985 is the minimum gain value signal870.

[0301] With the foregoing logic of FIGS. 8A-9C in mind, the input signalprocessing implemented by that logic can be better appreciated from FIG.10A. As will be appreciated by those skilled in the art, relevant stepsof the process shown in FIG. 10A are performed for each sample. Theprocess starts at step 1000, and advances to step 1003 at which a checkis performed to determine whether a change in inputs has occurred. If achange has occurred, such as the user ceasing use of a radio input andinstead selecting a DVD input, the process advances to step 1005. Atstep 1005, the selected inputs and the associated input level adjustersare enabled; the last minimum gain value associated with theprior-selected input device is stored in memory, and the appropriategain value for the newly selected device is loaded. As noted previously,in a presently preferred embodiment, the gain value is set either atmaximum or at the last valid value as stored in memory. As long as theinput device attached to the selected port has not been changed by theuser, the last valid value continues to represent the maximum amplitudesignal available from that input device.

[0302] If a change in inputs has not occurred, or after step 1005 hasexecuted, a check is then made at step 1010 to determine whether theinputs enabled in step 1005 are analog. If the check at step 1010 showsthe signals to be digital, the process branches to step 1015 where inputpackets are received and decoded, or input samples are taken. Theprocess then advances to step 1020 where the samples or packets aresupplied, in an exemplary embodiment, to FIFO buffers which permitasynchronous inputs to be transformed into a continuous rate output. Atstep 1025, the next samples are extracted from the FIFO buffers, afterwhich signal processing is performed at step 1030. Typical signalprocessing includes expansion and scaling of the signal represented bythe data packets. The branch then concludes and rejoins the sequencefrom the step 1010.

[0303] If the check at step 1010 showed that the inputs are analog, orfollowing completion of the signal processing at step 1030, the processadvances to step 1035 where input level adjusting is done, as furtherdescribed hereinafter in connection with FIGS. 10B and 10C. The processthen advances to step 1040, where the input signals are bandsplit andequalized, if desired for the particular implementation, and as furtherdescribed in connection with FIGS. 10D and 10E. The process then exitsat step 1045.

[0304] Referring next to FIGS. 10B and 10C, the input level adjust step1035 can be appreciated in greater detail. For clarity, two differentexemplary processes are discussed for implementing the input leveladjust process: a single loop process, shown in FIG. 10B, and a two loopprocess, shown in FIG. 10C.

[0305] The single loop input level adjust process of FIG. 10B begins atstep 1050, after which the process advances to step 1055 where a loop isbegun with the iteration of the loop being defined by how many inputsare processed. The number of inputs can vary over a wide range, and isidentified here as simply 1 to i. When the process loop begins, theinput signals are obtained and a loop timeout is initialized at step1060. The loop then advances to step 1065 to check whether the inputsignals are analog or digital. If the inputs are analog signals, theprocess advances to provide analog multiply and A/D conversion at step1070. If the check at step 1065 showed the signals were digital, adigital multiply is performed at step 1075. Whether the signals areanalog or digital, the process advances after either step 1070 or step1075 to step 1080, where a check for clipping is made. If no clipping isfound, the process advances to step 1085 to save the resulting 0 dBadjusted signal, after which the process loops back to step 1055.

[0306] If, however, clipping is detected at step 1080—which indicatesthat the signal has exceeded the 0 dB threshold and is not 0 dBadjusted, the process branches to step 1090, where a check is made todetermine whether a direct or an iterative calculate method is to beused to calculate gain. If the direct calculate method is to be used,the process advances to step 1095, where the minimum gain value iscalculated to avoid clipping. The gain can generally be represented asthe 0 dB signal amplitude divided by the input signal amplitude. Afterthe minimum gain is calculated, the process advances to step 1100 wherethe minimum gain value is updated. The process then returns to step 1065to reprocess the input sample with the updated minimum gain value 870.

[0307] If the check at step 1090 showed the minimum gain value was to bereduced by an iterative method, the process advances to step 1105, wherethe default minimum gain value is reduced by a predetermined fraction,where the fraction is between 0 and 1. The reduced gain value is thenstored as the new minimum gain value at step 1110. The process thenadvances to a “maximum loops” check at step 1115. The “maximum loops”check, which may not be required in all embodiments, establishes amaximum number of iterations which is permitted for a single sample. Byestablishing such a maximum the delay before accepting a new minimumgain value is potentially reduced; any further reduction may beperformed when the next sample is processed. If the loop timeout hasoccurred, the process advances to step 1085 and the 0 dB adjusted signalis saved. However, if the loop timeout has not yet occurred, the processloops back from step 1115 to step 1065, where another loop begins and afurther test for clipping is made, after which a further reduction maybe performed as necessary. It will be appreciated that the iterativeloop value takes longer, while the direct gain calculate method requiresgreater processing power. Regardless which method is used, once the newminimum gain value is calculated, the process ultimately saves the 0 dBadjusted signal (or its approximation, if the iterative process hasreached maximum loops without iterating sufficiently to achieve anaccurate 0 dB level during this sample) at step 1085 and returns to step1055 to process the next input. Once the appropriate number of loops hasbeen performed, the process exits at step 1160.

[0308] Although FIG. 10B shows a check being performed at step 1090, todetermine whether an iterative approach is used or whether a directcalculate method has been implemented, in most instances only oneapproach or the other will actually be implemented. In this instance, nocheck step 1090 is required. It has been shown here largely for ease ofexposition.

[0309] Referring next to FIG. 10C, the alternative of the two loop inputlevel adjust process can be better appreciated. A two loop processensures that all gain values applied to all inputs will be the same.With a single loop process, the minimum gain value changes when an inputclips, such that not all inputs will have the same gain value. By usingtwo loops, a minimum gain value which can be applied to all inputs isestablished by the first loop, after which that single minimum gainvalue can be applied to all inputs by means of the second loop. The twoloop approach thus offers slightly increased accuracy, although in mostinstances, the difference in output signal is difficult, if notimpossible, to detect. If the two loop process is selected, it starts atstep 1120, and advances to step 1125 where the first loop begins for 1to i inputs. Because the first loop is very similar to that shown inFIG. 10B, identical reference numerals will be used for identical steps,and will not be further described. The first loop of FIG. 10C differsfrom the loop of FIG. 10B in that the 0 dB adjusted signal is not storedyet, as shown at step 1085 in FIG. 10B. Instead, the process simplyreturns to the loop step 1125. Once the appropriate number of loops havebeen completed at step 1125, the process advances to the second loop atstep 1130. The loop advances to step 1135, to get the input and theassociated minimum gain value determined in the first loop. The processthen advances to step 1140, another check to determine if the input isanalog or digital. If the input is analog, an analog multiply and A/Dconversion is performed at step 1145, while if the input is digital adigital multiply is performed at step 1150. In either event, the processthen advances to step 1155 where the 0 dB signal for that input issaved. The loop then returns to step 1130 and, once the appropriatenumber of loops has been performed, the process exits at step 1160.

[0310] Referring next to FIG. 10D, the bandsplit and equalization step1040 from FIGS. 8C and 10A can be better appreciated. The process startsat step 1165, after which a loop begins at step 1170, with the number ofloops determined by the number of inputs. The loop advances to step1175, where a check is made whether a bandsplit is to be done for thatinput. If not, the process returns to step 1170 for processing of thenext channel. But if the channel being processed is to be bandsplit, theprocess advances from step 1175 to step 1177, which obtains theappropriate input signal to be bandsplit and provides it to step 1178,where the actual bandsplitting and scaling occurs. The process thenreturns to step 1170 for processing of the next channel. When allchannels have been processed, the process exits at step 1195.

[0311] Referring next to FIG. 10E, the bandsplit filter and scalingprocessor step 1178 from FIG. 10D can be better appreciated. The processstarts at step 1179, after which a loop begins at step 1180, with thenumber of loops to be processed at step 1180 is determined by the numberof bands the channel is to be split into. The loop advances to step1185, where the band output is calculated, after which scaling andequalization is performed and the results saved at step 1190. Theprocess then loops back to step 1180 for processing the next band. Oncethe requisite number of bands for that channel are processed, theprocess exits at step 1192.

[0312] Referring next to FIG. 11, exemplary logic for implementing theenvironmental sensor adjustments and noise extraction process can bebetter appreciated. A plurality of environmental inputs 470, which maycomprise a plurality of microphones or other sensors, are provided to aloop input processor 1200. In addition, the loop input processor getsreference signals from output signal processors 475 via signal bus 400B,and a series of control signals as shown in Table A via the control bus400A. The loop input processor provides acoustic loop balancing,negative loop feedback and signal conditioning functions. The loop inputprocessor 1200 provides as its outputs, typically, a fast referencepower estimator and a fast environment power estimator, both of whichare supplied to negative/positive loop comparison logic 1205. Theoutputs of the loop comparison logic 1205 provide both negative loopoutputs and positive loop outputs, which are fed back to loop inputprocessor 1200. While the loop input processor 1200 and loop comparisonlogic 1205 are both portions of the loop processor, signals are alsocommunicated between the loop processor and a noise processor. Inparticular, the control bus 400A provides a sensitivity signal tocorrection and conversion logic contained in noise processor 1210. Thecorrection and conversion logic 1210 provides a noise feedback signal tothe loop input processor 1200, while the positive loop outputs are alsoprovided to the noise processor 1210. The output of the noise processor1210 is a compander noise floor or volume control offset signal suppliedvia the control bus 400A.

[0313] The process implemented by the logic of FIG. 11 can be betterappreciated from FIG. 12, where the process begins at step 1215 and thenadvances to step 1220 for processing of environmental sensors andreference signals and negative loop feedback processing. It will beappreciated by those skilled in the art that the implementation of theenvironmental sensors and reference signals processing involves, in atleast some embodiments, processing of multiple bands and acoustic delaycompensation. The process then advances to step 1225, where a check ismade of whether a loop balance mode is active. If so, the processadvances to a loop balance step 1230. If not, the process advances tostep 1235 to process the positive/negative loops comparisons. Processsteps 1220, 1225, and 1230 are similar to the previously described loopinput processor 1200 and step 1235 is similar to the previouslydescribed negative and positive loop comparisons 1205. The process thenadvances to step 1245 for noise processing. After completion of step1245, or completion of step 1230, the process completes and exits atstep 1250.

[0314] Referring next to FIG. 13, the statistical engine 415 of FIG. 4may be better appreciated. The statistical engine inputs are providedfrom the control bus 400A, and comprise inputs 1 through n, which areprovided to statistical analysis histogram generators 1305A-n, whichprocess the historical data to generate histograms representative of theprior performance of the system. The histogram data outputs of thegenerators 1305A-n are provided to one or more statistical analyzers1310A-q. Signals are exchanged among the various analyzers 1310 via aninterstatistics engine control bus 1315, while the outputs of theanalyzers 1310A-q are statistical flags 1320A-q and statistical data1325A-q. In addition, the statistical analyzers 1310 and generators 1305receive as additional inputs statistical controls 650A-q. The analyzers1310A-q provide outputs in the form of histogram controls back to thegenerators 1305A-n. The statistical flags 1320, statistical data 1325,and statistical controls 650 are all communicated among the variouscomponents by means of the control bus 400A.

[0315] Referring next to FIG. 14, the process implemented by thestatistical engine logic of FIG. 13 can be better understood. Theprocess starts at step 1400, and advances to step 1405 where a loop isinitiated with one loop for each of the statistical engines. The loopadvances to step 1407 where a loop is initiated with one loop for eachof the histogram generators. The loop advances to step 1410, where theappropriate data is gathered from any desirable location in the system.Once the appropriate data is collected, the process advances to step1415, where the data is entered into the histogram generator appropriatefor this loop. When all the histogram generators have been processed,step 1407 advances to step 1417 where a loop is initiated with one loopfor each statistical analyzer. The loop then advances to step 1420 wherethe stored histogram data is analyzed. The results are then saved and,if required, appropriate statistical engine flags are set at step 1425,after which the histogram generators are updated at step 1430 and, whereappropriate, the histogram status is shared with other statisticalanalyzers. The process then loops back to step 1417 for processing ofadditional statistical analyzers. When all statistical analyzers havebeen processed, step 1417 loops back to step 1405 for processing ofadditional engines. Once the full complement of engines has beenprocessed, the process ends at step 1435.

[0316] Referring next to FIGS. 15 and 16, the transform engine 410 ofFIG. 4 may be better appreciated. The control bus 400A provides a seriesof inputs as shown in Table A, typically inputs from user interfaces,noise extractors and statistical engines, to one or more transformengines 1505A-t. Each of the transform engines then provides a series ofoutputs, which typically includes compander gain calculationcoefficients and kneepoints, volume control settings, and volume controlpre-mixer levels. In addition, an inter-transform control bus 1510shares information among the various transform engines 1505.

[0317] Referring next to FIG. 16, an example of the transform engine 410can be better appreciated. In this example, if the compander noise floorbecomes greater than the user minimum level, both the amount ofcompander compression (determined by outputs 2290) and the output volumelevel (determined by outputs 1640) are increased. The control bus 400Aprovides a user set minimum level to both a subtractor 1615 and selectmaximum value logic 1620. The subtractor 1615 receives as a second inputa compander noise floor signal from the bus 400A, which also serves asan input to the select maximum value logic 1620. The subtractor 1615provides only its zero or positive values to an adder 1625, which alsoreceives an input from the user volume control signal via the controlbus 400A. The output of the adder 1625 is provided to limit checks logic1630, which receives a second input from the select maximum value logic1620. The select maximum value logic also receives a control input fromthe inter-transform control bus 1510. The limit checks logic 1630receives a control signal from the control bus 400A, and provides anoutput to one or more input dynamic range transformation tables 1635 aswell as the inter-transform control bus 1510. The limit checks logic1630 also provides, as an additional output, a volume control setting1640. An input dynamic range signal is provided from the control bus400A to a table selector 1645, associated with the tables 1635. Thetables 1635 then provide compander gain calculation coefficients to thecontrol bus 400A.

[0318] Referring next to FIG. 17, the process implemented by thetransform engine 720 can be better appreciated. The process starts atstep 1700, and advances to step 1705 where the user interface data,noise extractor data, and statistical engine results are obtained. Theprocess then advances to a loop start at step 1710. The number of loopsis determined by the number of transform engines. The loop begins bygetting any required previous results from the appropriate transformengine at step 1715, followed by calculating new compression, volume,control and related variables at step 1720. The updated values are thensaved at step 1725, after which the process loops back to step 1710.After processing of the requisite number of loops, the process ends atstep 1730.

[0319] With reference to FIG. 18, the operation of a central powerestimator/mixer 455 (from FIG. 4) can be better appreciated. Asdiscussed in connection with FIG. 4, the power estimator mixer stage 455operates to permit multiple compander stages without altering relativespatial information or causing signal distortion between the variousbands or channels, and each separate listening environment may beprovided with a separate power estimator mixer. For purposes of exampleonly, FIG. 18 illustrates a power estimator/mixer for one channel whichis configured to process power estimates for multiple bands per channel.As shown in FIG. 18, one or more power estimate signals 1800A-n,typically exported local power estimates 2282 or global power estimates2281, are provided via the control bus 400A, with each power estimatesignal 1800A-n being supplied to associated input processing logic1805A-n. The input processing, which need not be implemented in allembodiments, can perform a plurality of functions, including a decoderfor signal reception (either a demodulator or packet disassembly), orlow pass filtering and/or a variable attack/release stage for reducedinter-channel and inter-band distortion. The outputs of the inputprocessing logic 1805A-n are each supplied to one or more powerestimator mixers 1810A-n, with the exact number being system dependent.For example, one mixer may be used for a pair of front speakers, whileanother is used for the rear speakers in a listening environment, wherethe two mixers communicate with each other. Each of the power estimatormixers 1810 in turn provides a single output signal representative ofthe power contained in the data signal being processed by thatparticular band of the channel. That output is supplied to associatedoutput processing logic 1815A-n, which in turn supplies a global powerestimate 2281 to the local power estimator/mixer in each compander asdescribed hereinafter. The output processing logic is not required inall implementations and, if used, will typically perform functionssimilar to the input processing stage, except that the output stage willperform encoding rather than decoding. It will be appreciated thatcompander implementations involving mixers configured for multiple bandsper channel will also typically have multiple bands per channel, inwhich case the global power estimates will be supplied to appropriateportions of the associated compander. It will further be appreciatedthat each of the input processing logic 1805A-n, power estimator mixers1810A-n, and output processing logic 1815A-n will receive a reset signalfrom the control bus 400A, at least for purposes of initialization.

[0320] Referring next to FIG. 19, the process by which the central powerestimators 455 develop global power estimates can be better appreciated.Again, FIG. 18 illustrates a central power estimator for a singlechannel with multiple bands, while FIG. 19 illustrates processing forall bands and all channels. The process starts at 1900, and advances tostep 1905 where all channel intermediate power estimates are obtainedand, if desired, optional input processing may be performed. The processthen advances to step 1910, where all power estimates for each of thebands within each channel (e.g., signals 1800A-n) are obtained, afterwhich a loop is entered at step 1915. As an alternative to the inputprocessing at step 1905, such input processing can also be performed atstep 1910. The loop begun at step 1915 repeats m times for m controlmixer algorithms. The loop starts at step 1920 by getting any previouscentral mixer algorithm results which are required for the algorithmexecuted by the loop. Step 1920 may also be used to perform inputprocessing. The process then advances to step 1925, where a specificcentral mixer algorithm is applied to the channel/band intermediatepower estimates together with any previous central mixer results. Then,at step 1930, optional output processing is performed and the resultsare saved for local post-power mixer use (i.e., the power estimatormixer which is local to the particular compander), and subsequent use bythe central mixer algorithm, after which the loop returns to step 1915to repeat as many times as necessary. After the requisite number ofloops have executed, the process advances to step 1935 and exits.

[0321] Referring next to FIG. 20A and the subsequent figures, thecompander stage 450 of FIG. 4 may be better appreciated. As will beunderstood by those skilled in the art, the compander stage is a keyelement in the overall system, although many other elements of thepresent system are novel. FIG. 20A provides a generalized view, with thesuccessive Figures providing greater detail of key elements. In FIG.20A, a plurality of input signals 2005A-n, which may in an exemplaryembodiment be 0 dB adjusted signals 860A-n, are supplied to one or morecompander groups 2010A-n, with a plurality of control signals as shownin Table A communicated bidirectionally between the compander groups2010 and the control bus 400A. The companded signals 2015A-n whichcomprise the outputs of the compander group 2010 are then provided tothe next stage via the system bus 400B.

[0322] With reference next to FIG. 20B, the compander groups 2010 ofFIG. 20A may be better understood. In particular, each compander group2010 typically comprises one or more companders 2020A-m, which may bemore typically thought of as adaptive dynamic companders in mostimplementations. One of the input signals 2005A-m is supplied to eachcompander 2020A-m within the compander group, and as shown in FIG. 20A aplurality of control signals (Table A) are communicated bidirectionallyvia the control bus 400A. The output signal 2025A-m, respectively, ofthe companders 2020A-m are combined in a signal combiner 2030, which istypically implemented as a mixer function. The signal combiner 2030receives control signals from the control bus 400A, as well. The outputof the signal combiner is provided to a soft clip stage 2035, alsodiscussed in greater detail hereinafter, and the output of the softclipstage 2035 is provided as the output of the compander group, or one ofoutput 2015A-n, e.g. 2015A. It will be appreciated that the function ofthe soft clip stage is similar to that discussed previously. In general,it functions to minimize output distortion caused by signal clipping forsignals that exceed the maximum signal level of the output stage. Forexample, if an input signal changes amplitude faster than the responsetime of the compander, the signal may be overamplified, and thereforeclip. The resulting output typically is detectable as distorted, and maybe perceived as unacceptable in some applications.

[0323] The soft clip stage functions to smooth the peaks of a signalthat exceeds the 0 dB level, typically in a non-linear manner. In analogdesigns, a soft clip function can be implemented with a diode clipper ora fast-acting compressor. However, when implemented as a DSP function,special techniques must be used because DSP implementations rely onsampling of the input signal. While analog signals are continuous,digital signals may be non-clipping in one sample, and fully clipped onthe next subsequent sample—with no intervening event or warning. Thus, aDSP implementation according to the present invention will preferablyprovide some headroom between the 0 dB level and the clipping thresholdof the system. When a signal amplitude crosses a predeterminedthreshold, for example the 0 dB level although other thresholds may beused, a routine is called which calculates the slew rate (dV/dt). Thatslew rate is then used to apply non-linear smoothing which keeps theoutput signal under the clipping threshold. Although more complicated,essentially the same predictive function is applied when the signalamplitude decreases below the 0 dB level. The function may beimplemented as a look-up table in at least some embodiments, althoughdirect calculation may also be used.

[0324] As will be appreciated from the foregoing, the dV/dt informationis used to determine how fast and how severely to smooth the inputsignal. Signals with high dV/dt values require more time for smoothingto avoid the abrupt flattening that causes clipping distortion. Toaccommodate this, they require a high soft-clip amplitude—i.e., moreheadroom above the 0 dB level—and the most severe amount of smoothing.Signals with lower dV/dt levels can be smoothed less severely, and mayalso be able to be smoothed over a longer period of time.

[0325] For implementations using a look-up table, when the signalamplitude exceeds 0 dB, the dV/dt information is used to select a tablehaving appropriately smoothed amplitude values. The look up table may beimplemented as a single monolithic table, or a plurality of discretetables; any reference herein to a look-up table contemplates bothimplementations. When the input signal amplitude returns to less thanthe 0 dB level, subsequent input samples can be used to compute the exitdV/dt, after which the appropriate look-up table values can be selected.Use of a delay buffer on the input signal may be required in at leastsome implementations to allow time for the exit dV/dt calculation. Inaddition, in some implementations the exit tables can simply be theentry tables read in reverse, thus eliminating the need for additionaltables.

[0326] Next, turning to FIG. 20C, the structure of an adaptive dynamiccompander 2020 may be better appreciated. The input signal 2005 (which,again, may be a 0 dB adjusted signal 860), is supplied from the systembus 400B to a half-wave signal processor 2040, a zero crossing ortimeout detector 2070, and a synchronizer block 2045. The half-wavesignal processor 2040, which is discussed in greater detail inconnection with FIGS. 21 and 22, receives control signals from thecontrol bus 400A, is enabled at zero-crossings or timeouts by the 0 dBadjusted signals, and calculates the appropriate gain for application atthat point in the signal, thus minimizing distortion. That is, if thenew gain is applied at the zero crossing, no detectable “glitch” in thesignal occurs, thus minimizing distortion. The half-wave signalprocessor provides as its output signal a “final gain” signal 2050, butintroduces a delay of typically a half cycle because of the need tocalculate the gain at the end of each half cycle. The final gain signalcomprises a second input to the synchronizer block 2045, which need notbe used in all instances but which functions to compensate for thesignal delay introduced by the half-wave processor. The synchronizerblock is helpful in improving transient response and also ensures thatthe gain calculated by the half-wave processor is applied on the correcthalf-cycle. The synchronizer block typically comprises buffer memories,and may be most easily implemented in a digital form. The synchronizerblock 2045 provides two output signals. The first is a delayed outputsignal 2055, while the second is a gain signal 2060. The delayed signal2055 is combined with the gain signal 2060 by multiplier 2065 to providethe output companded signal 2025 shown in FIG. 20B.

[0327] As discussed above, a desirable feature of a preferred companderaccording to the present invention is that it operates with very lowdistortion, relying in part on half-wave signal processing as showngenerally in FIG. 20C. With reference to FIGS. 21A-C, an exemplary formof synchronizer block 2045 and half-wave signal processor 2040 can bebetter appreciated. In general, the function of the synchronizer is toensure that the input signal is delayed by the same amount as the delayimposed by the half-wave signal processor 2040. An exemplaryimplementation can be appreciated from FIG. 21A, while the associatedinput and output waveforms can be better appreciated from FIGS. 21B and21C, respectively. The synchronizer block 2045 generally comprises asynchronizer 2045A and a gain buffer block 2045B, each so labeled andindicated by dashed lines in FIG. 21. The synchronizer portion 2045Areceives a sample clock signal 2105 from the control bus 400A, whichprovides the input to an input counter 2110 and an output counter 2115.The output of the input counter 2110 provides an input pointer to the“AIN” (address in) port of wave buffer 2125 and index buffer 2135, whilethe output of the output counter 2115 provides an output pointer to the“AOUT” (address out) port of the wave buffer 2125 and index buffer 2135.The input signal 2005, typically a 0 dB adjusted signal 860, provides a“DIN” (data in) input to the wave buffer 2125 and “DOUT” (data out)signal provides the delayed signal 2055. Wave buffer 2125 and indexbuffer 2135 are typically FIFO (first in first out) buffers as shown inFIG. 21A.

[0328] At the same time, the input signal 2005, typically the 0 dBadjusted signal 860, is also supplied to the half wave signal processor2040 and a zero cross detector 2070, which detects the zero crossingpoint of the input data stream. It also detects an input timeout whichoccurs if the input half wave length exceeds the length of thesynchronizer wave buffer 2125. The output of the zero cross detector2070 serves as the input to an increment index 2145 and enables the halfwave signal processor 2040 to calculated the final gain for the inputhalf wave that just ended. The output of the increment index 2145,typically a Modulo-“n” counter, provides the “DIN” (data in) input ofthe index buffer 2135 and also provides the input to an input pointergenerator 2155. The output of the input pointer generator 2155 providesthe “AIN” (address in) input to the gain buffer 2150. The output of theindex buffer 2135 serves as the input to an output pointer generator2160, which in turn provides the “AOUT” (address out) input to the gainbuffer 2150. The “DIN” (data in) input and “WCLK” write clock input ofthe gain buffer 2150 are supplied by the outputs of the half-wave signalprocessor 2040. It will be appreciated that the wave buffer and indexbuffer operate to continuously sequence the incoming data stream byestablishing a match between the “address in” of the incoming signal,and the “address out” of that same data suitably delayed to match thedelay imposed on the input signal 2005 by the half-wave signal processor2040. The result is the delayed signal 2055 and the gain 2060 aresynchronized. The half-wave signal processor 2040 also receives externaland global intermediate power estimates 2281, and exports local powerestimates 2282. It will also be appreciated that additional signaldelays may result from the use of half-wave signal processor 2040, inwhich case such delays may be compensated for by inserting appropriateadditional delay into the wave and index buffers.

[0329] The half-wave signal processor 2040 may be better appreciatedfrom FIG. 22. The half-wave signal processor may be thought of as activeonly at the end of a half-wave as indicated by zero crossings of theinput signal. The input signal 2005, typically the 0 dB adjusted signal860, is provided from the signal bus 400B to one or more half-cyclepower estimators 2270A-n. Here the input signal power is calculated on ahalf cycle basis, typically by calculating the peak, average, or RMSvalue of the half cycle input. These half cycle power estimatesfluctuate too rapidly for gain calculation use which would result insignal distortion. The half-cycle power estimators 2270A-n supplies aninput to one or more initial power estimators 2271A-m which smooth therapid fluctuations by varying amounts, typically by the use of lowpassfilters. Initial power estimators 2271A-m in turn provide initial powerestimator signals 2273 to one or more variable attack and releaseportions 2275A-v and control bus 400A for monitoring purposes. Thevariable attack and release portions 2275 control the rate at which thegain is allowed to change at the half-wave intervals. The variableattack and release portions 2275 typically work in series with the lesssmoothed initial power estimators 2271 and are perturbed to greater orlesser amounts by the less smoothed initial power estimators as will bediscussed in greater detail in connection with FIG. 23C. The resultinglocal intermediate power estimator signals 2279 are fed back to initialpower estimators 2271 and are also supplied to the control bus 400A formonitoring purposes, while various attack/release parameters aresupplied to the variable attack/release portions 2275A-v by the controlbus 400A.

[0330] The variable attack/release portions 2275A-v receive from thecontrol bus 400A a plurality of attack and release parameters 2274andoutputs intermediate power estimators signals 2279, which may in someimplementations have undergone a log conversion to allow easier signalmanipulation. It will be appreciated that log-converted signals may bemanipulated as linear signals, whereas more complicated calculations arerequired for signal which are not log converted. The log conversion,where desired, may be performed in any of several stages of the systemof the present invention. Regardless, the local intermediate powersignals 2279 are supplied to a multi-band/channel power estimator mixer2280, but also feeds power estimators signals 2279 back to the initialpower estimators 2271 and likewise provides the power estimators signalsto control bus 400A for use by other modules. Similarly, the mixer 2280receives as one or more inputs 2281 the external power and global powerestimates typically from other channels, bands, and central powerestimator mixers and also exports local power estimates 2282. Thefunction of the mixer 2280 is to maintain the relative amplituderelationship between channels, thus preserving spatial locationinformation. The mixer also helps to eliminate interband/channel phasecancellation and beating which can exist in at least some prior artdevices. The output of the mixer 2280, the final power estimator 2283,is an updated, or “post-mix” input power factor or value which issupplied to a Segmented Mapping Converter gain calculate stage 2285 andalso to the control bus 400A for monitoring purposes. The gain calculatestage 2285 operates on the final power estimator value 2283 to calculatethe final gain value 2050 required to generate the companded signal. Thegain calculate stage can also provide, as an output supplied to thecontrol bus 400A, for example, a log input power signal or selectedsegment indication 2287 for monitoring purposes. In addition, in someimplementations a half cycle peak value signal 2289 may be supplied bythe half-wave power estimator 2270 to the gain calculate stage 2285 toprovide look-ahead clip detection. In some implementations, thisarrangement may be used to determine if a signal will clip, in whichcase the peak value may be substituted to compute a non-clipping gainvalue. This may, in some instances, eliminate the need for apost-compander soft clip function. Other gain parameters 2290 aresupplied as controls to the gain calculate stage from the control bus400A.

[0331] It will be appreciated that, for a number of applications of thepresent invention, not all of the stages shown in FIG. 22 are required.Thus, for many implementations of the present invention, theattack/release stage is not required. Likewise, in other implementationsthe multi-channel power estimator mixer is not required.

[0332] Referring next to FIGS. 23A-23C, an exemplary embodiment of ahalf-cycle power estimator 2270 and an initial power estimator 2271,each respectively shown by a dashed line box, can be better appreciated.FIG. 23B shows the relationship between the K and K′ lowpass filtercoefficients used in the initial power estimators of FIG. 23A and anincreasing number of samples per half cycle (which corresponds to adecreasing frequency.) As the number of samples in a given half wavedecreases, it is less likely that the peak input value will be sampledon every half cycle. This beating between the input and the sample ratecauses fluctuations in the half cycle peak value which increase withincreasing frequency (less samples per half cycle). The fluctuations canbe eliminated by use of a lowpass filter however an adequate filterresults in very slow response times to low frequency inputs. The secondissue is that the effective sample rate of the digital lowpass filtersdecreases with input frequency since there are less half cycles per unitof time. This causes the corner frequency of the lowpass filters todecrease slowing the low frequency response even more. To counteractthese effects, the lowpass filter coefficients of the initial powerestimators can be made to vary with input frequency to produce aconstant response Fc over all input frequencies as shown in FIG. 23B.Alternatively, the coefficients can be made to vary so that the responsetime is different for different input frequencies, for example, instantresponse for low frequencies and slower response for high frequencies.FIG. 23C shows a series of exemplary waveforms for the inputs andoutputs of the stage of FIG. 23A and the action of variable attack andrelease portions 2275A. In successive sequence, each signal smooths thefluctuations of the previous signal. The intermediate power estimate2355 is a serial combination of the slow initial power estimate 2360 ofFIG. 23A and a variable attack and release portion described more fullyin connection with FIGS. 24B and 24C. The slow initial power estimate isresponsible for the smooth appearance of signal 2355 while the variableattack/release is responsible for the rapid perturbation of the signal.This provides a smooth power estimate for low distortion compandingwhile still being able to provide excellent tracking of the input powerwhich is required for reduction or elimination of compander gainundershoots and overshoots and possible output clipping.

[0333] Referring particularly to FIG. 23A, the input signal 2005,typically 0 dB adjusted signal 860, is supplied to the zero crossingdetector 2070 and to a peak detector 2305. The output of the zerocrossing detector 2070 provides as its output to inverting reset inputsof the peak detector 2305 and sample counter 2310, and also to thenon-inverting clock inputs of a “number of samples” register 2315 andpeak value register 2320. The sample clock 2105 serves as a count inputto a sample counter 2310, and provides its output to the register 2315.Similarly, the output of the peak detector 2305 serves as the input tothe peak value register 2320. This allows the half cycle power estimatorsample length counter and peak detector value to transfer to the initialpower estimator at an input signal zero crossing to start the half wavesignal processor. The sample length counter and peak detector are thenreset for the next half cycle.

[0334] The peak value in 2320, with appropriate filtering, provides areasonable representation of the power of the 0 dB adjusted signal. Theoutput of the “number of samples” register 2315 can readily be seen tobe the number of samples taken of a signal between zero crossings. In anexemplary arrangement, the output of the samples register 2315 issupplied to a look-up table 2325 which relates the number of samples tofilter pole parameters K and K′ and equalization EQ. The lookup table2325 may be configured for, for example, 16K samples/second. The EQvalue from the lookup table 2325 is supplied to a multiplier 2330, whereit is combined with the peak value signal 2335 from the peak valueregister 2320. This compensates for the effective lowering of higherfrequency peak half cycle values since it is less likely that the peakinput value will be sampled on every half cycle. The output of theequalizer 2330 is supplied to a low pass filter/intermediate powerestimator 2340, which also receives the K filter pole parameter from thelookup table 2325. Similarly, the K′ filter pole parameter is suppliedfrom the lookup table 2325 to a low pass filter/intermediate powerestimator with external feedback 2345. The output of the low pass filter2340 serves as a fast initial power estimator signal 2350, and issupplied as an output from the stage as well as an input to the filters2340 and 2345. The filter 2345 also receives as an input an intermediatepower estimate signal 2355 from the associated variable attack andrelease modules 2275A-v, which can be better appreciated from FIGS.24A-C, and provides as its output a slow initial power estimate 2360.The values of K and K′ basically define the corner frequencies of thelow pass filters 2340 and 2345, such that varying values of K and K′causes the filters 2340 and 2345 to be variable low pass filters able tocompensate for the issues discussed previously in FIG. 23B.

[0335] Referring next to FIGS. 24A-24G, which show in greater detail thevariable attack and release portion 2275, FIG. 24A can be seen to showlogic for a generalized variable attack and release portion 2275 whichmay include multiple linear or non-linear segments. FIG. 24B shows anexample of a single segment variable attack and release portion whichcan be either linear or non-linear. FIG. 24C shows a preferredembodiment of a single segment nonlinear attack and release module witha filter coefficient K″. FIG. 24D shows the relationship between variousfixed attack and release values of K″ ranging from 0 (slowest attack andrelease) to 1 (fastest attack/release) for the preferred embodiment ofFIG. 24C. FIG. 24E shows a plot of one segment linear transforms forvarying values of B where K″ is defined by the relationship BΔ. Thedelta variable is the absolute value of the difference (typically linearor log based) between a fast initial power estimator 2350 and a slowinitial power estimator 2360. The difference represents the errorbetween the fast tracking (more realistic) and the slow tracking (forlow distortion) filter outputs. This error can be applied to a varietyof math functions, in this case a simple linear equation, to producevarying degrees of perturbation of the slow filter to more accuratelytrack the input signal while still providing low distortion. It will beappreciated that fast initial power estimates dominate at the upperrange of the plot, while slow initial power estimates dominate at thelower range of the plot. FIG. 24F shows a plot of non-linear singlesegment transforms where K″ is defined by the equation K″=αΔ²+βΔ+λ,while FIG. 24G shows the variable attack and release output waveformswhich result from various values of K″.

[0336] With specific reference first to FIG. 24A, a plurality of inputs2400A-n (which may, for example, be the initial power estimates 2273, ornoise floor value 5417, or volume control offsets 5419) are supplied toa plurality of math processors 2405A-n. The math processors, which maybe broadly thought of as a comparison stage, include both linear andnon-linear processing functions, can exchange calculation results overthe Δ Value bus, and also receive feedback inputs from a feedback bus2415. The math processors output functional expressions on a mathcontrol bus 2420A and a math function bus 2420B. The math processors2405 perform calculations on the external, feedback, and Δ value inputsto provide the variables for the segment parameter selectors 2425A-m,segment processing transforms 2430A1-mq and tracking adjusting filters2427A-o which are provided via the math function bus 2420B. The mathprocessors can also contain state machines and logic, the results ofwhich are provided to the segment parameter selectors 2425A-m andtracking adjusting filters 2427A-o via math control bus 2420A. Segmentparameter selectors 2425 are used to facilitate the approximation ofcomplex segment processing functions by the use of a plurality ofsegments of simpler functions. Redundant hardware or computations can beeliminated by the ability of each segment parameter selector to provideinputs to many segment processing transforms. The segment parameterselectors 2425 may be broadly thought of a first stage that operates onmath processor outputs which may be modified by a second stageresponsive to user preference signals, feedback signals, etc. Aplurality of segment parameter selectors and associated segmentprocessing transforms can be used for processing different math functioninputs or implementing different attack and release behaviors, forexample linear difference processing and logarithmic differenceprocessing. The segment parameter selectors 2425 also receive as inputsvarious attack/release parameters 2274, typically internal configurationparameters 640 or gain calculate parameters 2290, which can be suppliedeither from the user interface 405 or the transform engine 410, both ofwhich were discussed in connection with FIG. 4. The outputs of thesegment parameter selectors 2425A-m provide a plurality of coefficientsto an associated plurality of segment processing transforms 2430; thus,segment parameter selector 2425A has associated therewith segmentprocessing transforms 2430A1-2430A-n, while segment parameter selector2425 m has associated therewith segment processing transform 2430 m 1through 2430 mq. The segment processing transforms 2430 each receive asan additional input the function bus outputs from the math processors2405A-m via the function bus 2420B.

[0337] The output of each of the segment processing transforms 2430,which may be broadly thought of as a transform stage, is a segmentfilter coefficient, which is supplied to an attack/release segmentcombiner 2440, which may be broadly thought of as a combiner stage. Herethe segment filter coefficients can be blended or the appropriate oneselected for use on the various tracking adjusting filters. Theattack/release segment combiner 2440 generates a plurality of finalfilter coefficients and supplies them to the tracking adjusting filters2427A-o, which may be broadly thought of as a tracking filter and can bea low pass filter similar to the low pass filters 2340 and 2345(discussed in greater detail in connection with FIGS. 24B-24F), or canbe more complicated such as will be discussed in connection with FIGS.57A and 57B. The output of the filters 2427 are supplied as the localintermediate power estimates 2279, or noise signals 5425 and 5427, andare also supplied to the math processors 2405A-n via the feedback bus2415, as discussed previously. The operation of the math processors2405A-m, segment selectors 2425A-m, transforms 2430, combiners 2435 and2440 and filter 2430 will each be discussed in further detailhereinafter.

[0338] With specific reference to FIG. 24B, a single segment transformcan be better appreciated. External inputs 2400, in this example thefast initial power estimator signal 2350 and the slow initial powerestimator signal 2360, are provided to the math processor 2405, whichprovides both a set of attack and release parameters (bus 2420A) anddifference and difference squared variable values (bus 2420B) to asegment parameter selector 2425. The segment parameter selector 2425also receives inputs 640 consisting of compander slope and user selectsinformation. The selector 2425 outputs a plurality of segment filtercoefficients to a segment processing transform 2430, which also receivesthe difference variables via the bus 2420B and in turn generates a finalfilter coefficient K″. The final filter coefficient K″ is supplied tothe tracking adjusting filter 2427, which also receives the fast initialpower estimator signal 2350 slow initial power estimator value 2360. Thetracking adjusting filter 2427 outputs an intermediate power estimate2355, which is an element of bus 2279.

[0339] Next referring to FIG. 24C, there is shown therein a preferredembodiment of a single segment attack and release module with anonlinear filter coefficient K″. The configuration of elements from FIG.24B is shown in dashed lines in FIG. 24C, including the math processorportion 2405, the segment parameter selector 2425, the segmentprocessing transform 2430, and the tracking adjusting filter 2427. Inparticular, the fast initial power estimate 2350 and slow initial powerestimate 2360 (e.g., FIG. 23A) are provided to a difference calculation2455. The difference 2455 is provided to logic 2462 which generates anattack and release logic signal for addressing a lookup table 2460.Lookup table 2460 is also addressed by compander slope 2290, which isconverted to a positive/negative slope logic signal by element 2464, andone or more user interface internal configuration parameters 640. Thelookup table generates the α, β, and λ values in response to the inputs,and supplies them to, respectively, multipliers 2465 and 2470 and adder2475. The remaining inputs to the multipliers 2465 and 2470 are providedby the process Δ variables block 2463, which can perform additionalcomputations (such as the difference squared) and can also performnumeric conversions on any input values or calculation results (e.g.linear to logarithmic conversions). The outputs of the multipliers 2465and 2470 are combined with the value in adder 2475 to establish thevalue K″. The results from the adder 2475 are supplied to trackingadjusting filter 2427, in this example to a multiplier 2480 and to asubtractor 2485 which determines the values of 1−K″. The value of 1−K″is then combined in a multiplier 2490 with the slow initial powerestimate 2360. The value K″ is combined with the fast initial powerestimate 2350 in the multiplier 2480, the output of which is added tothe output of the multiplier 2490 in adder 2495 to yield theintermediate power estimate 2355 (part of intermediate power estimators2279).

[0340] The transform characteristics of attack and release portion canthus be seen to depend in large measure on the values used inestablishing the value of K″. Shown in the table below are variousconsiderations for combinations of fast or slow attack and fast,moderate or slow release. The results of the varying attack and releasevalues on the intermediate power estimates 2279 can be seen graphicallyfrom FIG. 24G. The compress and expand tables shows how differentK″=αΔ²+βΔ+λ equations are used for compressing or expanding signals anddifferent user selectable attack/release responses. The slope=+/−M,+attack, −release, and user select signal can be found in FIG. 24Cexample. Compress Table Compress +Attack −Release (Slope = +M) (Δ =Positive) (Δ = Negative) Comment User Select 0 K“ = 0Δ² + K“ = 1Δ² +Maximum linear 1Δ + 0 0Δ + 0 attack, good (fast linear (moderatecompromise between attack) non-linear) fast transient response &“breathing”, for compression ratios: 1:1 to 10:1 User Select 1 K“ =0Δ² + K“ = 0Δ² + Linear maximum 1Δ + 0 1Δ + 0 attack & release, (fastlinear (fast linear provides good attack) release) transient response,good for data signal capture, but not for voice. Pumping and warbleproblems. User Select 2 K“ = 0Δ² + K“ = 0Δ² + Slowest response, 0Δ + 0Δ0Δ + 0 lowest distortion, (slowest (slowest worst transient attack)release) response User Select 3 K“ = 0Δ² + K“ = 0Δ² + Peak detect,never >0 1Δ + 1 0Δ + 0 dB, no crest factor (fastest (slowest problem.Good for attack) release) compression ratios >10:1. Worst releaseresponse which can cause under amplification of weak signals.

[0341] Expand Table Expand +Attack −Release (Slope = −M) (Δ = Positive)(Δ = Negative) Comment User Select K“ = 0Δ² + K“ = 0Δ² + 0Δ + 0Preferred, causes 0 to 3 1/16Δ + 0 (slowest release) lowest amount of(slow linear “breathing effect attack) distortion

[0342] Referring next to FIG. 25A, the operation of the local powerestimator mixer 2280 may be better appreciated. The local intermediatepower estimates 2279 or initial power estimates 2273 from each localchannel/band input power estimate may, if desired, be supplied to alocal input processing function 2505 and export processing functions2510A-x. The local input processing function also receives controlinputs from the bus 400A, which likewise supplies control signals to theexport processing functions 2510A-x as well as an import processingfunction 2515 and a local power estimator mixer function 2520. Theimport processing function operates on the external powers estimates andglobal power estimates 2281 and provides its output to the local powerestimator mixer function 2520, which receives the output of the localinput processing function 2505 as another input. The local inputprocessing function 2505, export processing functions 2510A-x, andimport processing function 2515 are each optional, depending on theparticular implementation desired; one or all may be eliminated inspecific implementations of the present invention, yet provide morerobust functionality when implemented. Examples of local inputprocessing are combining a plurality of local power estimates into asingle local estimate or selecting the appropriate input such as thelargest or smallest input value. An example of export processing is themodulation of a carrier wave for transmission to a central or anotherlocal power estimator. Import processing would then demodulate thesignal. The primary output of the mixer 2280 is supplied by the localpower estimator mixer 2520 and forms the final power estimate 2283. Thelocal power estimator mixer 2520 combines the internal and externalpower estimates or selects the appropriate one.

[0343]FIG. 25B shows an example of a local power estimator with onelocal power estimate and three external power estimates. A local powerestimate signal 2279 is supplied to power estimator mixer 2520 and isalso provided as an exported power estimate signal 2282. There is nooptional export processing in this example. External power estimates2281A,B,C are provided to import processing logic 2515 where the maximumexternal power estimate value is selected, divided by a factor of twoand supplied to power estimator mixer 2520. Power estimator mixer 2520then selects the larger of the local or import processed external powerestimate for use as a final power estimate signal 2283. The importprocessed power estimate signal is divided by two so that if all bandsor channels are close to the same power estimate value, the local powerestimate value has priority and thus reduces distortion.

[0344] Referring next to FIG. 26A, a generic example of a SegmentedMapping Converter to perform the gain calculate 2285 portion of FIG. 22can be better appreciated. The gain calculate portion shown in FIG. 26Acan be seen to include a plurality of gain calculation blocks 2600A-n,each of which receives an associated input power signal 2620A-n, adaisy-chained calculate enable signal 2633A-n, which may be broadlythought of as command enable signals, and gain calculate parameters2290, which may be broadly thought of as transform parameters. The inputpower signals 2620A-n can be received from the final power estimators2283, half cycle peak values 2289, or the intermediate power estimates2279. Each gain calculate block 2600A-n comprises segmented gaincalculate logic block 2605A-n, which may be broadly thought of assegmented transform processors, where each such logic block 2605A-nreceives both the associated input power signal, e.g. 2620A, theassociated calculate enable signal, e.g. 2633A, and gain calculationparameters 2290. The output of the segmented gain calculate logic blocks2605A-n are selected or combined in select or combine logic block 2610A,the calculated gain output (also referred to as a select/combine output)of which is then provided to test logic 2615A. This allows for parallelgain computations, with the gains being combined or the appropriate gainbeing selected in block 2610A, the combining or selection parametersbeing provided by gain calculate parameter 2290. Each gain calculateblock 2600A-n generates as an output signal an initial gain signal2640A-n, which can be seen from gain calculate block 2600A to be takenfrom the associated test logic, e.g. 2615A. Each test logic block2615A-n can also be seen to provide the calculate enable signal 2633B-nto the next successive gain calculate block 2600B-n. Since there is atleast one gain calculate block, the first block 2600A has as a defaultthe Calculate Enable 2633A always true or enable. Test logic block 2615Atests the Calculated Gain signal provided by select or combine block2610A and if useable (test passes) passes the 2610A Calculated Gain tobecome the Initial Gain 2640A and sets Calculate Enable 2633B to “false”to disable any further calculations by subsequent 2600B-n Gain Calculateblocks. If test logic block 2615A determines that the gain signalprovided by select or combine block 2610A is not useable (test fails)then the Calculated Gain is not passed to Initial Gain 2640A andcalculate enable 2633B is set to “true” to enable subsequent gaincalculation blocks. The Calculate Enable can also include additionalinformation, such as a command, that the segmented gain calculate blocks2605 can use in gain calculation. The 2615 test block testing allows forserial gain calculations to be performed until a useable gain has beencalculated. Note that the last gain calculate block need not includetest block 2615. The initial gain signals 2640A-n are made available forsubsequent processing as final gain signal 2050.

[0345] While FIG. 26A shows a generic form of gain calculate block, FIG.26B illustrates a particular embodiment in which a serial implementationis used, where the gain calculation is combined with a predictive clipdetection and gain correction function. In particular, gain calculateblock 2600A is provided with the calculate enable signal 2633A, set to atrue default setting since it is the first gain calculate block, and thepower estimate signal 2620, in this example final power estimate 2283.Those signals are used by segmented gain calculate block 2605A andselect or combine block 2610A, together with gain calculate parameters2290, to perform the gain calculation portion of the gain calculateblock 2600A, which results in the calculated gain value. The calculatedgain value is then tested (as shown at 2615A) to determine whether theresulting gain will result in signal clipping (compander output exceedsthe 0 dB level) by multiplying the half cycle peak value 2289 by thecalculated gain and comparing the result to the 0 dB level. If no, thecalculated gain is acceptable and is provided as initial gain 2640A andultimately as the final gain signal 2050 for the stage. If, however, thecomputed gain multiplied by the half cycle peak value exceeds the 0 dBlevel, the clipping indication and enable is provided by the CalculateEnable 2633B signal to the next gain calculate stage 2600B, which alsoreceives the half cycle peak value signal 2289 and the gain calculateparameters 2290. The gain calculate block 2600B then recalculates acorrected gain which will not produce a compander output that exceedsthe 0 dB level, which is provided as initial gain 2640B and ultimatelyprovided as the final gain 2050. It can be appreciated that, while FIG.26B shows only two gain calculate stages 2600, multiple such stages maybe used.

[0346] An alternative parallel implementation can be realized bycomputing the gain based on the final power estimate and half cycleestimates in parallel segmented gain calculate blocks 2605A,B and thenselecting the non-clipping gain in the select or combine block 2610A. Inthis case, test 2615A is not required since the correct gain waspreviously selected by block 2610A.

[0347] Referring next to FIG. 26C, the segmented gain calculate block2605 of FIG. 26A may be better appreciated. In particular, FIG. 26Cshows the gain segment variables 2662A, typically for the first gainsegment block 2645A being the input power signal 2620, being provided toa gain segment block 2645A, which may be broadly thought of as a segmentselection processor, and in particular to a numeric conversions logicblock 2650 within the block 2645A. The numeric conversions block may beused to perform, for example, a linear-to-logarithmic conversion tosimplify subsequent calculation, although such conversion is notrequired in all instances. The calculate enable signal 2633 is providedto gain segment block enable logic 2647 which allows overall processingin this block to occur and to gain segment selector 2655 for use indetermining which gain segment/variable processor 2660A-n to use. Theoutput of the numeric conversions logic 2650, if used, is provided to again segment selector 2655, which also receives as an input the gaincalculate parameters 2290, which includes definition of the segmentboundaries. While complicated non-linear gain calculations can be used,it is sometimes desirable to divide the input power 2620 range intosegments, where each segment uses a less complicated calculation, toemulate the more complicated calculation. The gain segment selectorlogic 2655 basically divides the input power range into appropriatesegments, and selects and passes data to the gain segment/variableprocessor 2660A-n appropriate for the current input power 2620 value.Each of the processors 2660A-n receives the gain calculate parameters2290, and from the gain segment selector 2655 receives data and asegment select signal. In addition, the processors each receive a gainsegment coefficients signal 2630. Typically, the first gain segmentblock 2645A does not require any gain segment coefficients 2630A. Eachprocessor thereupon develops a gain segment variable 2662 and a gainsegment coefficient 2630, which can be provided to subsequent gainsegment blocks 2645B-n. The use of multiple levels of gain segmentblocks allows for serial segmenting of the input power range. Forexample, the first gain segment block may divide the input power rangeinto two segments, one if the input power is greater than 0 dB and oneif less, and may do the selection using linear input values while thesecond gain segment block may further segment the less than 0 dB segmentinto a middle and lower segment using logarithmic converted input powervalues, the input power being passed through the first gain segmentblock via gain segment variables 2662. Following processing of thevarious gain segment blocks 2645A-n, the final gain segment variables2662 n and final gain segment coefficients 2630 n are provided to a gaintransform calculation 2665, which may be broadly thought of as atransform processor, which outputs an initial gain value 2640 or, ifappropriate, a final gain value 2050.

[0348] The segmented gain calculation function can be better appreciatedfrom FIG. 26D, in which an exemplary implementation of the segmentedgain calculate function of FIG. 26C is shown using one gain segmentblock of four segments. This exemplary implementation can also be usedin gain calculate blocks 2600A and 2600B of FIG. 26B, the calculateenable 2633A and 2633B signals being used to select and enable theproper gain calculation. The first time through the gain segment block(e.g. block 2600A), the calculate enable 2633 is always enabled andindicates non-clipping and there are no coefficients 2630. The initialstep is to perform a linear to logarithmic input conversion of the finalpower estimates 2283 as shown at numeric conversions block 2650, sincethis reduces the complexity of subsequent calculations. The gain segmentselector 2655 then subtracts the compander 0 dB offset 2290 from the loginput power 2287. The compander 0 dB level, which is based on a filteredlong term average of the peak input power, can be different from theinput level adjuster 0 dB level, which is based on peak input levels,due to the crest factor of the input source signal. If the resultantvalue is negative, then headroom segment 2660A is selected, causing themaximum output level to be limited. If, however, the gain segmentselector resultant value is less than a lower kneepoint as specified bythe compander gain calculate parameters 2290, then lower segment 2660Cis selected, causing the maximum gain to be limited. Otherwise, themiddle segment 2660B is selected, allowing normal compander operation.Note that the clip segment 2660D will never be selected due to calculateenable 2633 indicating a non-clipping state.

[0349] After the appropriate segment has been executed by the gainsegment coefficient and variable processors 2660, the gain transformcalculation 2665 computes the actual gain, in this example by use of asimple line equation, providing the calculated gain, initial gain 2640or final gain 2050.

[0350] If predictive clipping is implemented, as in FIG. 26B, theclipping test would be performed and if clipping were to occur, gaincalculate block 2600A would set the calculate enable 2633B signal toclipping and enable gain calculate block 2600B. The half cycle peakvalue 2289 is converted to a logarithmic value 2287 by numericconversion block 2650. The gain segment selector 2655 then subtracts thecompander 0 dB offset 2290 from the log input power 2287. Since thecalculate enable signal indicates clipping, the clip segment 2660D isselected, the output used by gain transform calculation 2665 to computea non-clipping gain value.

[0351] As an alternative implementation of FIG. 26B, the example of FIG.26D can be split into two segmented gain calculate blocks. Gaincalculate block 2600A would use a three segment segmented gain calculateblock using gain segment coefficient and variable processor blocks2660A,B,C, and Gain Calculate block 2600B would use a one segmentsegmented gain calculate block using gain segment coefficient andvariable processor block 2660D. The calculate enable signal 2633 thendoes not require clipping information but additional software code orhardware is required.

[0352] Referring next to FIG. 26E, how each segment calculates gain maybe better appreciated. In particular, the gain 2050 is shown on thevertical axis, ranging from 80 dB gain to −60 dB attenuation, and thelog input power 2287 is along the horizontal axis. The hashed area showsthe maximum gain or attenuation that can typically be realized. Typicaloperation will be within this area as shown by the solid linesrepresenting the Y=MX+B and Y=M (lower kneepoint)+B equations. Theheadroom segment acts as a limiter as shown by the line sloping down tothe left of the compander 0 dB point. The middle segment performscompanding and is realized by the equation, gain=compander slope M (loginput power)+B. The lower segment limits the maximum gain by fixing thelog input power value to that of the lower kneepoint log value. The clipsegment forces the output to the compander 0 dB level by using infinitecompression. This is shown as the maximum upper boundary of the middlesegment. Conversion of the input power to a logarithmic value allows theuse of simple line equations.

[0353]FIG. 26F shows the output power resulting from various amounts ofcompanding. The log input power 2287 is shown along the horizontal axisand the output power (log[companded signal 2015]) is shown along thevertical axis. The headroom segment is for inputs larger than thecompander 0 dB point. They are limited to the maximum 0 dB output levelshown by the horizontal line to the right of this point. Inputs lessthan the compander 0 dB point but greater than the lower kneepoint arecompressed or expanded as shown in the middle segment. Inputs less thanthe lower kneepoint have their gain fixed at the lower kneepoint gain asshown in the lower segment. The clip segment 2630B forces the input to 0dB output level and would appear as a line on the horizontal axis.

[0354] Referring next to FIG. 26G, illustrates how the log input power2287 and MX+B linear gain transform may be used to access the final gainvalue 2050 from a lookup table. First, compander slope M (one of thevalues 2290 provided by the transform engine 410) is multiplied by loginput power 2287. This results in positive values for compression andnegative values for expansion. To convert this to a value suitable fortable lookup, typically a positive only value, an offset value B isadded. B is typically an offset value to the unity gain 0 dB gain valuein the table. The gain table contents are typically linear multipliervalues for compander gain cell use. The log to linear gaintransformation is accomplished in the lookup table.

[0355] Turning next to FIGS. 26H-J, an example is shown in FIG. 26H of again calculation using a non-linear gain transform function, in thiscase a high order polynomial equation. Unlike the multiple linearsegments used in FIG. 26B, only a single nonlinear segment is necessaryin this example because of the curve fitting possible with the nonlinearsegment. The polynomial variable is set to the log input power value andthe polynomial coefficients are calculated by the gain segmentcoefficient processor 2660 according to the amount of compression orexpansion required, as specified by the compander gain calculateparameters 2290. These are then used in the gain transform calculation2665. It will be appreciated that gain segment processor 2660 couldinclude a plurality of equations, and the gain transform calculation2665 could be implemented to select one of such plurality, or to providecurve fitting. If a selection method is implemented, the selection couldbe any of a variety of choices, including selecting the minimum ormaximum. FIG. 26I shows how the polynomial equation calculates the gainsmoothly as the log input level changes. This curvilinear approachavoids the distortion that can occur at abrupt segment boundaries. Inthis example, four pseudo segments are smoothly realized. A linearsection allows compensating for the input signal crest factor. FIG. 26Jshows how the non-linear polynomial gain calculation smoothly varies thecompander output level for several compression and expansion settings.

[0356] Having described the logic by which the compander and its variouselements are implemented, the process of operation for the compander canbe better appreciated by FIG. 27A et seq. Referring first to FIG. 27A,the overall operation of a generalized form of compander function as atshown 370 of FIG. 3B, and in FIGS. 20A-C, may be better appreciated. Theprocess begins at step 2705, and advances to optional step 2710 where acheck is made to determine if the system is in setup mode. If so, theprocess bypasses the remaining steps and advances to exit 2715. Iffalse, as will usually be the case, the process advances to step 2720,where a loop is initiated for 1-n channels. The loop at step 2720 callsa sub-loop at step 2725, for 1-m bands. For each of the m bands, theprocess advances to step 2730 where the input signal for that band andchannel is obtained. The process then advances to a further sub-loop atstep 2735, for 1-g companders, where each loop includes a half-wavesignal processing step 2740, followed by updating the synchronizerinputs and getting the synchronizer outputs at step 2745. Then, at step2750, an alternative check to 2710 can be made to determine whether thesystem is in setup mode, or what may be thought of as setup mode check2; if so, the loop jumps to its end by returning to step 2735. If not,the process advances to a gain cell routine at step 2755, after whichthat loop completes and returns to step 2735. This alternative setupmode check 2 will allow the channels and bands to complete the halfwavesignal processor 2740 and synchronizer block 2745 so that when setupmode exits, a smoother return to normal companding will result. However,additional processing is required compared to setup mode check 1. Bothsetup check mode 1 and 2 are optional, the disadvantage being that allof the compander steps will execute unnecessarily since in setup mode acalibration signal is typically output instead of the compander output.

[0357] When all companders have been processed, step 2735 proceeds tothe second optional setup mode check 2 test at step 2757 to determinewhether the system is in setup mode 2. If so, the process jumps to step2725; if not, the process advances to step 2760, where the signalcombiner function is performed which combines all of the compander gaincell results from compander loop 2735. The process then advances to thesoft clip step 2765, after which the process loops back to step 2725 forthe next band. The signal combiner is not required unless there is morethan one compander per band. The soft clip function may also not berequired in all implementations. The process repeats for the remainingbands, after which the process returns to step 2720 for the remainingchannels and their associated bands and companders. Ultimately, afterthe n^(th) channel is processed, the process exits at step 2715.

[0358] Taking next FIGS. 27B and 27C together, the operation of a splitcompander arrangement is shown in process flow diagram form. Forsimplicity, like elements with respect to FIG. 27A have been shown withlike reference numerals. In general, the split compander process ofFIGS. 27B-C differs from the compander process of FIG. 27A in that thecentral power estimator step 725 of FIGS. 7 and 19 occurs in the middleof half-wave signal processing step 2740 of FIG. 27A. This guaranteesthat the central power estimator results are computed with the initial,intermediate, or final power estimates from all companders for thecurrent input signal sample so that the central power estimator resultsare in sync with the input samples. Computing the central powerestimator results before or after the compander shown in FIG. 27A willtypically result in a one sample delay, which is typically acceptable tothe listener. The process begins at step 2705 and advances to step 2710,where a check is made to determine whether the system is in setup mode.If so, the system bypasses the remainder of the process show in FIGS.27A and 27B, and advances to an exit at step 2715. If not (as willusually be the case), the system branches to step 2720, where a loop isbegun for 1 through n channels. That loop in turn calls another loop atstep 2725, for one through m bands per channel.

[0359] In turn, that loop advances to step 2730, where the band andchannel signal input are retrieved and the process then advances to step2735, where another loop is begun to process each of one through gcompanders. From step 2735, the process advances to step 2737, where acheck is made to determine whether the sample indicates a zero crossing.If so, the process advances to step 2739 where the half cycle flag isset for the particular channel and band being processed. The initialpower estimator process is then performed at step 2742, where the halfcycle power estimator is also reset, followed by processing the variableattack and release intermediate power estimates at step 2747. Theprocess then advances to the half-cycle power estimator process at step2749.

[0360] If the test at step 2737 had determined that the sample beingprocessed was not a zero crossing, the process would have advanced tostep 2753, where a timeout test is performed to ensure that a zerocrossing has occurred within a predetermined period. If the timeout testshows that no zero crossing has occurred within the required time, theprocess branches to step 2739 just as discussed above. However, in mostcases the timeout test will be false, and the process will advanceimmediately to step 2749. From step 2749, the process advances to step2735 and the remaining companders are processed. Once all the compandersfor a given band have been processed, the system returns to step 2725,and the next bands in sequence are processed. Eventually all the bandsfor a given channel will have been processed, after which the processreturns to step 2720; at this point the bands and companders associatedwith the next channel are processed. Eventually all channels will havebeen processed, and the overall process advances from step 2720 to step2768 where the central power estimator mixer process is called.Following completion of the central power estimator mixer, the processadvances to step 2770, where a loop is begun for 1 through n channels.That loop in turn calls another loop at step 2774, for one through mbands per channel.

[0361] In turn, that loop advances to step 2778, where the band andchannel signal input are retrieved and the process then advances to step2780, where another loop is begun to process each of one through gcompanders. From step 2780, the process advances to step 2782 where acheck is performed to see if the half flag is set for this channel/band.The half flag would have been set if a zero crossing or timeout occurredat step 2737 or 2753 for this band or channel, to indicate at this pointthat the half wave signal processing should be completed. If the halfflag is not set, then the process proceeds to step 2745. If the halfflag is set, then the process proceeds to the reset half flag step 2784,local post power estimator mixer step 2786, gain calculate step 2788,and then update synchronizer inputs and get synchronizer outputs step2745. Then, at step 2790, an alternative check to 2710 can be made todetermine whether the system is in setup mode, or what may be thought ofas setup mode check 2; if so, the loop jumps to its end by returning tostep 2780. If not, the process advances to a gain cell routine at step2755, after which that loop completes and returns to step 2780.

[0362] When all companders have been processed, step 2780 proceeds tothe second optional setup mode check 2 test at step 2795 to determinewhether the system is in setup mode 2. If so, the process jumps to step2774; if not, the process advances to step 2760, where the signalcombiner function is performed which combines all of the compander gaincell results from compander loop 2780. The process then advances to thesoft clip step 2765, after which the process loops back to step 2774 forthe next band. The signal combiner is not required unless there is morethan one compander per band. The soft clip function may also not berequired in all implementations. The process repeats for the remainingbands, after which the process returns to step 2770 for the remainingchannels and their associated bands and companders. Ultimately, afterthe n^(th) channel is processed, the process exits at step 2715.

[0363] Referring next to FIG. 28, the half-wave signal processing showngenerally at step 2740 in FIG. 27A can be better appreciated. Theprocess starts at step 2800, and advances to step 2805 where a check ismade to determine whether the signal is at a zero crossing. If not, theprocess advances to a timeout check at step 2810, in case an unexpectedevent has caused the signal to be lost such that no zero crossings occuror the length of a half cycle exceeds the length of the synchronizerblock 2045 buffers. If the check at step 2805 is true—that is, thesignal is at a zero crossing, or if a timeout has occurred as determinedby the check at step 2810, the process branches to step 2815, where theinitial power estimator values are processed, and the half cycle powerestimates are reset. The initial power estimator values may be betterunderstood from FIG. 30, discussed hereinafter. The process thenadvances to step 2820, where the variable attack/release values areprocessed and the intermediate power estimates are generated, asdiscussed in connection with FIG. 31 et seq., hereinafter. The processthen advances to step 2825, where the multi-band/channel power estimatesare generated. Then, at step 2830, the gain values are calculated. Ifthe check at step 2810 was false, or upon completion of the gaincalculation at step 2830, the process advances to step 2835, where thehalf-cycle power estimator process is performed, as better explained inconnection with FIG. 29. The process then exits at step 2840.

[0364] As noted above, FIG. 29 shows in greater detail the process ofthe half cycle power estimator, which begins at step 2900 and advancesto step 2905 where a loop is begun for each of a plurality of half cyclepower estimators, for example 1-e. The loop includes step 2910, where ahalf cycle power estimate is generated; the power estimate may be basedon any convenient indicia, including peak, average, RMS, and so on. Theloop then advances to step 2915, where the number of samples per halfcycle counter is incremented. The loop then returns to step 2905 forprocessing of the next half cycle power estimator; after the last suchpower estimator is processed, the process exits at step 2920.

[0365] Referring next to FIG. 30, the initial power estimates of step2815 (FIG. 28) may be better appreciated. The process begins at step3000, and advances to step 3005 where all half cycle power estimatorvalues and all values for the number of samples per half cycle count aresaved. Then, at step 3010, the half cycle power estimates and the numberof samples per half cycle are reset for the next half cycle, after whichthe process advances to step 3015. At step 3015, a loop is called foreach of 1-m half cycle power estimators, with each loop including, atstep 3020, generating an equalization value for the half cycle powerestimator using the count value from the number of samples per halfcycle, followed by applying that equalization value to the half cyclepower estimator value at step 3025, and saving the equalized half cyclepower estimator value at step 3030 after which the loop returns to step3015 for processing of the next power estimator. Following completion ofthe loop for each of the half cycle power estimators, the processadvances from step 3015 to step 3035.

[0366] Step 3035 calls a second loop for processing 1-p Initial PowerEstimators, which begins at step 3040 by using (step 3005) the countvalue for the number of samples per half cycle to generate algorithmparameters for the initial power estimators. Then, at step 3045, theprocess gets any required previous initial power estimates, equalizedhalf cycle power estimates, and intermediate power estimates and appliesthem to the initial power estimator algorithm. The initial powerestimates are then saved at step 3050, after which the loop returns tostep 3035 for processing of the next power estimator. Once all initialpower estimators are processed, the process exits at step 3055.

[0367] Referring next to FIG. 31, the variable attack and releaseprocess, shown as step 2820 in FIG. 28 and associated with the exemplarysystem discussed in connection with FIGS. 24A-24G, may be betterappreciated. The process starts at step 3100, and advances to the mathprocessors function (further described in connection with FIG. 32,hereinafter) at step 3105. Following the math operations, the processadvances to the segment processor step at 3110 (described in greaterdetail in connection with FIG. 33), followed by the segment combinerstep 3115 for the attack and release function (described in more detailin connection with FIG. 34.) Thereafter, the tracking adjuster filtersstep is performed at 3120 (treated in more detail in connection withFIG. 35), after which the process exits at 3125.

[0368] With reference next to FIG. 32, the math processors step of FIG.30 can be better appreciated. The process starts at step 3200 andadvances to step 3205, where the initial power estimates and feedbackinputs are obtained. The process then advances to step 3210, where aloop is called to process each of a plurality of 1 to m math processors.The loop includes step 3215, where previous math processor results areobtained, followed at step 3220 by applying the selected math processoralgorithms to the initial power estimates and feedback inputs as well asprevious math processor results. Math processors typically perform mathfunctions such as but not limited to input differences, squares, cubes,etc. of the difference, absolute value operations, as well as statemachine and logic functions such as but not limited to input differencepositive or negative, or input difference polarity change indication.The results are then saved at step 3225 for use by the segmentprocessors and subsequent math processors, after which the loop returnsto step 3210. After each of the math processors has been processed, theprocess exits at step 3230.

[0369] Referring next to FIG. 33, the processing step involving thesegment processors (step 3110 in FIG. 31) can be better appreciated. Theprocess starts at step 3300 and advances to step 3305, where a loop iscalled for processing 1 through s segment processors. The loop initiallygets, at step 3310, the required math processor results together withthe user interface controls and compander operating parameters, whichare then used to generate, at step 3315, the segment parameters andapply the same to the segment processing transform. The process thenadvances to step 3320 where the coefficients are saved for use by thecombiners. The loop then returns to step 3305, and after processing thelast segment processor, the process exits at step 3325.

[0370] The attack and release segment combiner process, shown in FIG. 31at step 3115, can be better appreciated from FIG. 34. The attack andrelease segment combiner process begins at step 3400 and advances to afirst loop at step 3405, for 1 to I combiner levels. The loop from step3405 calls a second loop initiated at step 3410 for each of 1 to c(I)combiners, where the loop includes getting, at step 3415, theappropriate coefficients (stored at step 3320). The coefficients arethen used in applying the combiner algorithm for the particularcombiner, after which the derived coefficients are stored at step 3425for use with the tracking filters and subsequent combiners. After eachcombiner has been processed, the loop called at 3410 returns to step3405, and the next combiner level is processed. Once all combiners ofeach of the combiner levels have been processed, the process exits atstep 3430.

[0371] The tracking adjuster filters process, shown summarily in FIG. 31as step 3120, can be better appreciated from FIG. 35. The process startsat step 3500 and advances to step 3505 where a tracking filter loop iscalled, for processing 1 through t tracking filters. The loop includesgetting, at step 3510, the appropriate input power estimates, feedbackvalues, previously calculated intermediate power estimates, and (fromstep 3425) the appropriate coefficients. The tracking filter algorithmis then applied at step 3515, after which the resulting intermediatepower estimates and feedback values are saved at step 3520. Trackingfilters for companders typically implement some form of variable lowpass filter while other uses, such as for noise compensation, may useintegrators, low pass filters, and non-linear filters separately or incombination. Following completion of all loops for 1 through t trackingfilters, the process exits at step 3525.

[0372] With reference next to FIG. 36, the process of the local postpower estimator/mixer as previously discussed in connection with FIG. 25may be better appreciated. It will be appreciated that the local postpower estimator mixer is shown simply in FIG. 28 at step 2825. The localpost power estimator/mixer process initiates at step 3600 and advancesto step 3605 where the appropriate power estimates (either localintermediate or initial) are processed and/or combined, and the resultsare saved for the use with the local power estimator mixer algorithm andexport processing. The process advances to step 3610 where the powerestimates are processed and/or combined for export to the central powerestimator mixer and, as appropriate for the particular implementation,other bands or channels.

[0373] The process then advances to step 3615 where the external powerestimates from other bands/channels and/or global power estimates fromthe central power estimator mixer are obtained, for use at step 3620 inapplying the local power estimator mixer algorithm to produce the finalpower estimate. The final power estimate is then saved at step 3625 forgain calculation use, after which the process exits at step 3630.

[0374] Referring next to FIG. 37A, a generic example of a SegmentedMapping Converter to perform the gain calculate process, shown at step2830 in FIG. 28, may be better understood. In particular, the processstarts at step 3700 and advances to step 3705, where a loop is calledfor 1 through g gain calculators. The loop includes the step of apply asegmented gain calculate algorithm, shown at step 3710, which may bebroadly thought of as a segment transform processor algorithm. After thelast relevant segmented gain calculator has been processed, the loopreturns and the system advances to step 3715 where the gain calculatorresults are selected or combined to form the computed gain value (alsoreferred to as a select/combine output value). The output is then testedat steps 3720 and 3725, for example to ascertain whether clipping occurs(e.g., exceeds the 0 dB level) by multiplying the half cycle peak valueby the calculated gain. If the test fails (e.g. signal clipping willresult), the process branches to step 3730, where the test results aresaved, which may be broadly thought of as the command enable signalgeneration, and the process loops back to step 3705 to compute a moreappropriate gain value (e.g. one that will not result in signalclipping). The test results are typically the equivalent of thecalculate enable 2633. Once a gain is determined which passes the test3720 (e.g. no clipping occurs), the process advances from the testconducted at step 3725 to step 3733, where the gain value is saved, andthen exits at step 3735. If a synchronizer 2045 is used, then the savestep 3733 may not be required since typically the gain will be stored inthe synchronizer.

[0375] A presently preferred embodiment of a gain calculate process canbe better appreciated from the flow diagram of FIG. 37B, which isparticularly suited to a parallel processing implementation. The processstarts at step 3740 and advances simultaneously along two branches: tostep 3745 where a first segmented gain calculator algorithm is appliedto the final power estimate and to step 3750, where a second segmentedgain calculator algorithm is applied to the half cycle peak value. Asnoted for step 3710, above, the segmented gain calculator algorithm ofsteps 3745 and 3750 will be discussed in greater detail in connectionwith FIG. 38. Both branches then supply their results to a select orcombine step 3765, in this example a select step, where the firstsegmented gain calculator algorithm result is applied to the half cyclepeak value. If the result exceeds the 0 dB level, the second segmentedgain calculate algorithm is selected to avoid clipping. Otherwise, thefirst segmented gain calculation result is selected. Once theappropriate segmented gain calculator algorithm result is selected, itis saved at step 3755 and the process exits at step 3770.

[0376] Referring next to FIG. 37C, an alternative serial embodiment ofgain calculator process is shown, wherein both predictive clip detectionand gain correction are implemented. The process starts at step 3775 andadvances to step 3780 where the segmented gain calculator algorithm isapplied to the final power estimate to calculate compander gain. In thisexample, since there is only one segmented gain calculator 3710, loop3705 and select or combine step 3715 are not required. Then, at step3785, the half-cycle peak value is multiplied by the compander gain togenerate the peak half-cycle value. At step 3790, the result from step3785 is compared to a predetermined clipping threshold, typically the 0dB level. Steps 3785 and 3790 comprise the apply testing steps 3720 and3725. If the peak half-cycle value exceeds the clipping threshold, thenthe test results are saved (clipping=true) at step 3793 and thecompander gain calculation is repeated at step 3795 by applying thesegmented gain calculator algorithm to the half-cycle peak value andtest results. Either after getting a false result at step 3790, orcompleting the gain recalculation at step 3795, the compander gain issaved at step 3796 and the process exits at step 3798. In this example,since the second pass through gain calculate 2830 for gain recalculationguarantees an acceptable gain value, loop 3705, select or combine 3715,test application 3720, and test 3725 are not required.

[0377] As with the gain calculate algorithm steps of FIGS. 37A and 37B,the segmented gain calculate process of step 3710 can be betterunderstood from the following discussion of FIG. 38. The process startsat step 3800 and advances to step 3805, where, depending on theimplementation, test results 3730, or a loop from step 3830, the gainsegment input variables (typically one of the input final powerestimates or half cycle peak value) or previous gain segment variablesresults are obtained. Optionally, the values from step 3805 undergo anumeric conversion such as a linear to log conversion at step 3810,followed by retrieval of test results 3730 at step 3815, for use in thegain segment selector process at step 3820. The gain segment selector,which may be broadly thought of as a segment selection processor,determines which segment includes the input value and accommodates thetest results. The process then advances to step3822, where the gainsegment coefficients are obtained, typically the result of a previoussegment loop 3805 through 3830, the first time through the loop thecoefficients typically being null values. The process then advances tostep 3825, where the gain segment coefficients and variables for theparticular segment are generated. The process then advances to step3830, where a check is made to determine whether the segment beingprocessed is the last segment. If not, the process loops back to step3805 and the next segment is processed as above. Once the last segmenthas been processed the check at step 3830 yields a true result and theresults from the gain segment processor step 3825 are used to generate again value at step 3835, which may be broadly thought of as a transformprocessor step. The process then exits at step 3840.

[0378] Turning next to FIG. 39, the process step shown at 2745 in FIG.27A and 27C can be better appreciated. The process, which updates thesynchronizer inputs and gets the synchronizer outputs, starts at step3900 and advances to step 3905, where the input signal sample, typicallyadjusted to 0 dB, is retrieved and placed in the synchronizer FIFObuffer. Then, at step 3910, the delayed signal sample, also typicallyadjusted to 0 dB, is extracted from the synchronizer FIFO buffer so thatthe sample can be used by the compander gain cell. At step 3915, thegain value calculated for the extracted signal sample is obtained fromthe gain buffer and made available to the compander for compander gaincell use. The process then exits at step 3920.

[0379] The soft clip process, shown in summary form at step 2765 of FIG.27A, can be better appreciated from FIGS. 40A, 40B and 40C. As noted inconnection with FIG. 27A, the purpose of the soft clip process is tomanage the distortion that can result when the input signal and the gainresult in an output signal above an acceptable threshold. While FIG. 40Ashows the process for application of a soft clip, examples of thevarious types of signals which might lead to clipping are shown on theleft side of FIG. 40B, while the associated signal resulting afterapplication of the soft clip process is shown on the right side of FIG.40B. It will also be appreciated that the “entry” portion of the softclip process differs from the “exit” portion of the process, where thepositive or negative values greater than the positive or negativeclipping thresholds separates the entry from the exit portions.

[0380] The process of FIG. 40A starts at step 4000 and advances to step4005, where the current input signal value is retrieved. A test is madeat step 4010 to determine whether the sample is within the clip region.This is typically done by comparing the magnitude of the input signaland a clipping threshold or by comparing the input signal to an upperthreshold and a lower threshold. If so, the process branches to the“clip entry” path, which begins at step 4015, where a check is made todetermine whether this is the first sample within the clip region. Ifso, the clip length counter is reset, the clip event counter isincremented and a clip signal may be generated for use by otherprocesses at step 4020, followed by calculating the slope of the curvedefined by the previous and current samples—or dV/dt, at step 4025. Theclip event counter may be reset at any time. After calculation of dV/dt,or if the test at step 4015 turns out false (which means simply that aprior sample was in the clip region), the calculated dV/dt is used togenerate smooth clip value for the current sample during the “entry”portion of the process, typically by use of a look-up table. The clipcounter and dV/dt values can be used as pointers to address a lookuptable that contains the smoothed clip output value. Different dV/dtvalues can be used to access sections of the lookup table appropriatefor the amount of smoothing required. As shown in FIG. 40B, the fastdV/dt inputs require more amplitude to smooth the signal versus the slowdV/dt inputs and different lookup tables (or portions thereof) used togenerate the different outputs shown on the right side of FIG. 40B.Alternatively, the clip counter and dV/dt values can be used to directlycalculate the smoothed clip output values. The dV/dt value may also becontinually calculated between samples while in the clip regions. Theclip length counter is then incremented at step 4035, followed at step4040 by placing the smooth clip signal value generated in step 4030 intothe FIFO buffer. The process then advances to step 4041 where thedelayed signal value is extracted from the FIFO buffer for furtherprocessing, after which the process exits at step 4043.

[0381] If, however, the test at step 4010 yields a false result, theprocess advances to a test at step 4045 to determine whether the currentinput signal value obtained at step 4005 is exiting the clip region.Exiting the clip region occurs when the current input sample value isnot in the positive or negative clip regions or transitions from thecurrent polarity clip region to the opposite polarity clip region. Ifso, the process branches to step 4050 and the output dV/dt isdetermined, using the last smooth clip signal value less the currentinput signal value. The process then advances to step 4055, where a loopis called for N samples, typically the lesser of the clip length dividedby two or the FIFO buffer depth for example although other values can beused. The loop advances to step 4060 where the dV/dt value and cliplength divided by two or the FIFO buffer depth (whichever is smaller)are used to generate the exit smooth clip value, typically from a lookuptable or computation, and load it into the appropriate FIFO location foreach sample. Once the last sample is processed, the loop at 4055 exitsand executes the check at step 4063. This check tests for the case wherethe current input sample value transitions from the current polarityclip region to the opposite polarity clip region and if true branches tostep 4020 to generate “entry” smoothed clip output values. If false,then the process then returns to the same point as though the test atstep 4045 had returned a false (i.e., no clipping and not exiting aclip, or what will be the most common process in response to a sample),and advances to step 4065 where the current input signal—either theactual sample or adjusted for clipping—is input into the FIFO buffer.The process then advances to step 4041 as described above, followed byan exit at step 4043.

[0382] Referring to FIG. 40C, the entry and exit process may be betterunderstood. The input signal waveform can be observed from right toleft, with the initial portion not exceeding either the positive upperor negative lower clipping thresholds. The samples which form thedigital representation of the signal are indicated by X, circles and,later, squares. Eventually the signal enters the clipping region byexceeding the clip threshold, the circles indicating the “entry”smoothed clip output values generated by the softclip algorithm.Eventually the sample drops below the clip threshold at which point theoutput dV/dt is determined and the FIFO is backfilled with “exit”smoothed clip output values (squares), in this example for half of thelength of the FIFO buffer to produce smooth “entry” and “exit” curve.While bipolar operation is shown, the same method applies to unipolarinputs.

[0383] The various elements which comprise the compander portion of thepresent invention can thus be understood. Following the companderfunction, the volume control 445 portion of the system of FIG. 4 can bebetter appreciated. A generalized view of the multi-module volumecontrol arrangement in accordance with the present invention can bebetter appreciated from FIG. 41A, in which the signal input and outputto the volume control stages is provided via the signal bus 400B,comprising particularly multi-channel/multi-band signals 4125A through4125 n and calibration signals 5010A-n, which are supplied to volumecontrol and pre-mixer modules 445A-v. The various modules 445 eachreceive control signals from the control bus 400A, while the output ofthe volume control stage is provided as one or more single/multi-bandsignals 4130A-4130 n, which are then distributed via the signal bus400B. Thus FIG. 41A illustrates the use of multiple volume control andpre-mixers to implement one level of a multi-band and/or multi-channelvolume control.

[0384] Turning to FIG. 41B, the volume control and pre-mixer 445 of FIG.41A may be appreciated in greater detail. The signal bus 400B provides acalibration signal 5010 together with the multi-channel/multi-bandsignals 4125A-4125 n. The multi-channel/multi-band and calibrationsignals are provided to a pre-processor 4100 which also receives controlsignals from the control bus 400A in the form of volume controlpre-mixer levels 4210A through 4210 n. Typically the preprocessor 4100is a signal mixer with input scaling. The resulting signal is providedas the “A” input to a signal selector function 4105 while thecalibration signal 5010 provides a “B” input thereto. In addition, thesignal selector function 4105 also receives a calibrate 640 signal toselect the A or B inputs. The output of the signal selector function4105 is provided to a volume cell 4110, which applies to the inputsignal a volume setting control signal 1640 and outputs asingle/multi-band channel signal 4130A.

[0385] Turning next to FIGS. 42A-42D, examples of a variety of volumecontrol configurations may be better appreciated. FIG. 42A shows asingle band/channel input plus volume control, where the signal 4125A issupplied to the volume control function 4110, resulting in a singleband/channel output signal 4130A. FIG. 42B shows three bands of a singlechannel for the input signal 4125A, requiring a pre-mix function 4100together with the volume control 4110 function in the volumecontrol/premixer 445. FIG. 42C shows two channels, each with threebands, as inputs, with a single band, single channel out. Thus, twothree-input premixers 4100 are provided, one for each channel, togetherwith a premix-volume control 4110. The outputs of the pre-mix volumecontrol are provided to a mixer 4100, which is then supplied to thevolume control function as described for FIG. 42B. It will beappreciated that each pre-mix and volume control combination can beconfigured from the function 445 described previously.

[0386]FIG. 42D shows a volume control configuration in which twochannels groups, each with three bands per channel, are scaled and mixedwith volume control applied to yield a single channel of three bands.Like FIG. 42C, the configuration is a matrix of volume control/pre-mixfunctions 445. Thus, each channel group is supplied to a premix volumecontrol 4110, with appropriate pairings (first with first, second withsecond, third with third, in the example shown) of the premix volumecontrol outputs to the premix portion 4100 of a second function 445,followed by a volume control function 4110, yielding three outputsignals. It will be appreciated that many alternative configurations arepossible, and these examples are provided only to show a framework forthe manner in which such signals might be combined.

[0387] Referring next to FIG. 43A, the output signal processing function(shown as 475 in FIG. 4) is represented in block diagram form. Thesignal bus 400B provides a plurality of single/multi-band channelsignals 4330A through 4330 n, each of which is provided to the outputsignal processor block 475, and in particular to an associated channelprocessing function 4300-A4300 n. The output signals from each of theblocks 4300A is a channel band group 4335A-4335 n, and is provided to anassociated band group output processor 4305A through 4305 n (discussedin greater detail in connection with FIG. 43B). Control signals for eachof the functions 4300 and 4305 are provided from the control bus 400A.Control signals to block 4300 typically consist of bandsplit and scalingparameters and to block 4305 typically consist of signal combiner,softclip, bandsplit filter/scaling and output conversion parameters andare listed in table A. In turn, each of the output processors 4305A-4305n provide a channel reference out signal 4310A-4310 n to the control bus400A, and also provide analog or digital outputs for their respectivegroup. The output signals include signals 480, 482 and 485 in FIG. 4.

[0388] Next referring to FIG. 43B, the band group output processor 4305can be better appreciated. Each of the signals 4335A-n, which may be aplurality of signals, is provided to an associated one of a plurality offunction blocks 4315A-n which serve as signal combiners, softclip, andbandsplit filters. The control bus 400A typically provides signalcombiner, softclip, bandsplit filter/scaling and output conversionparameters to the function blocks 4315A-n as well as output conversionblocks 4320A-n. The output signals from blocks 4315A-n are band groupsignals 4340A-n, and are supplied to respective ones of the outputconversion blocks 4320A-n (better explained in connection with FIG. 44.)Each of the output conversion blocks provides an associated outputconversion reference 4345A-n, all of which are provided to a referencecombiner block 4350 which may be broadly thought of as a referencesignal generator. The output conversion blocks 4320A-n also provideappropriate analog or digital output signals 480, 482 and 485. Thereference combiner 4350 provides the channel reference output signal4310 for that output processor.

[0389] As noted above, FIG. 44 describes in greater detail the outputconversion block 4320. As shown in FIG. 43B, the band group signal4340A, which may be a plurality of signals are provided as the signalinputs to an input summation processor 4400, which also receives controlsignals from control bus 400A as discussed previously. The InputSummation Processor 4400 provides as one output the output conversionsreference signal 4345, and in addition provides an output signal toanalog output processing block 4405 and to digital output processingblock 4430. The control bus 400A also provides command signals to theoutput processing blocks 4405 and 4430. The analog output processingblock 4405 provides discrete sound outputs 480 as well as output 485 forother analog signal-accepting devices. The digital output processingblock 4430 provides digital outputs 482, typically in the form ofpackets or other digital format.

[0390] Turning next to FIGS. 45A-45G, exemplary arrangements for singleand multiple channel outputs are shown. A simple single band channelarrangement is shown in FIG. 45A, where channel input signal 4330 issupplied as an input signal to output conversion block 4320 as therequisite component of the band group output processor 4305 without anychannel processing 4300. The output conversion block provides a singlespeaker output 480 together with a reference output 4345 that may beused as a channel reference output 4310.

[0391] A single band channel with speaker equalization is shown in FIG.45B. Input signal 4330 is supplied to band group output processing block4305, the first element being a bandsplit filter which comprises part ofthe block 4315. The bandsplit filter 4315, typically similar to process1178 of FIG. 10E, operates to provide speaker equalization, and outputsa plurality of signals to the output conversion block 4320, which inturn output the speaker output 480 as well as the reference output 4345that may be used as a channel reference output 4310.

[0392] A somewhat more robust single band channel is shown in FIG. 45C,this time with tri-amped output and speaker equalization combined. Theinput signal 4330 is supplied to a bandsplit filter and scaling block,which is part of channel processor 4300 of FIG. 43A and typicallysimilar to process 1178 of FIG. 10E. A plurality of tone controls issupplied by control bus 400A as bandsplit scaling coefficients. Thebandsplit filter 4300 outputs bass, midrange and treble signals to atrio of bandsplit filters, parts of blocks 4315 of band group outputprocessor 4305. Each of the bandsplit filters 4315 operates to providespeaker equalization, and provides a plurality of bandsplit outputs toassociated output conversion blocks 4320, each of which in turn providesa speaker output 480 and a reference output 4345A-C. The referenceoutputs 4345A-C are then combined in the reference combiner 4350, whichoutputs the channel reference output 4310.

[0393] Referring next to FIG. 45D, an exemplary arrangement of a singleband channel with tone control can be better understood. Thesingle/multiband channel input signal 4330 is provided to a channelprocessor 4300, which also receives a plurality on tone control signals(bandsplit scaling coefficients) via the bus 400A. The channel processor4300 particularly includes, for this example, the bandsplitfilter/scaling function, typically similar to process 1178 of FIG. 10E,,which output treble, bass and midrange signals. The three band signalsare combined in a signal combiner (part of 4315), in this example asimple adder, the result of which is supplied to a soft clip portion ofthe functional block 4315. The soft clip output is provided to theoutput conversion block 4320 as discussed previously, which in turnprovides the speaker output 480 and reference output4345 which may beused as a channel reference output 4310.

[0394] Turning next to FIG. 45E, an example is shown of a single bandchannel with tone controls and three multi-amped speakers. The signal4330 is supplied to a bandsplit filter and scaling block, part of thechannel processor 4300. The bandsplit filter block 4315 outputs in thisexample, bass and sub-woofer outputs directly to output conversionblocks 4320 (part of the output processor 4305), while the treble andmid-range signals are combined in an adder which comprises the signalcombiner part of 4315. As with the other examples, the output conversionblocks 4320 each outputs a speaker output signal 480 as well as areference signal 4345A-C. The reference signals are combined in areference combiner 4350, which in turn outputs a channel referenceoutput 4310.

[0395]FIG. 45F illustrates an exemplary multi-band channelimplementation, in which a three band signal 4330 is supplied directly(i.e. no channel processing 4300) to the band group output processor4305, and more specifically to a signal combiner (part of 4315), fromwhich the band group signal is supplied to an output conversion block4320. The output conversion block then output the appropriate speakeroutput 480, as well as a reference output 4345 which may be used as achannel reference 4310.

[0396] A more robust implementation, suitable for use with a multibandchannel and including a sub-woofer, can be seen in FIG. 45G. A multibandchannel 4330 supplies a treble, midrange and bass signal to a channelprocessor stage 4300, and the bass signal particularly to a bandsplitfilter portion of 4300. The bandsplit filter portion splits the basssignal into a bass and sub-woofer signals. The treble and mid-rangesignals are combined in a signal combiner which is part of a block 4315,and the resulting band group is supplied to the output conversion block4320. Similarly, the bass and sub-woofer signals are provided toassociated output conversion blocks 4320. As with the other examples,the reference outputs 4345A-C of the conversion blocks 4320 are eachprovided to a reference combiner 4350. The output conversion blocks alsogenerate a speaker output 480, while the reference combiner 4350generates a channel reference output signal 4310.

[0397] The foregoing examples provide some indication of the versatilityof the functional aspects of the output processing portion of thepresent invention. It will be appreciated that the previous exampledesigns can be expanded to multiple channels.

[0398] Referring next to FIG. 46, the volume control block 373 processmay be better appreciated which generally corresponds to the logic ofthe multi-module volume control and pre-mixer modules 445 as previouslydiscussed in connection with FIG. 41A. The process starts at step 4620and advances to step 4625, which calls a loop for one to m channel andbands. For each channel and band, the loop advances to a volume controland pre-processor process 4630, after which the result of the process4630 is saved at step 4635. The loop then returns to the step 4625 forprocessing of the next band or channel. After all of the bands andchannels have been processed, the loop returns and exits at step 4640.

[0399] With reference next to FIG. 47, an exemplary version of thevolume control and pre-processor process 4630 may be better appreciated.the process starts at step 4700, and advances to a check at step 4705,where a determination can be made whether the system is in setup mode.In those instances where the system is in setup mode, such as duringcalibration, the process branches to get step 4710, where a calibrationsignal input sample is obtained. If, as will usually be the case, thetest at step 4705 confirms that the system is not in setup mode, theprocess advances to step 4715, where the relevant compander outputs orother signal values, for example the previous level volume control 4635results, are obtained. Those outputs are then applied at step 4720 to apre-processor combiner/mixer algorithm, which may for example simply addthe signals together or may split or scale the particular input signals.Whether as the result of step 4720 or getting the calibration signalinput sample of step 4710, the process then advances to step 4725 wherethe volume control setting is obtained, typically from the transformengine 720 or user interface 360, and applied to the signal, typicallyby use of a multiplier, after which the process exits at step 4730.

[0400]FIG. 48 illustrates an exemplary form of robust output signalprocessing, which generally corresponds to the logic of FIG. 43B, andwhich covers many of the output processing permutations for producingboth an output signal and a reference signal for use in noisecompensation or other purposes, such as an intelligent user interface.The process starts at step 4800, then advances to what was generallyshown as the output signal processor step 375 in FIG. 3B, and morespecifically to step 4805 where a loop is called for one to n channels.The loop begun at step 4805 calls another loop at step 4810 for one to mbands; the loop may generally be thought of as the channel processingportion 4811, in this example a bandsplit function. The loop advances tostep 4815, where a test is made to determine whether the signal is to besplit into sub-bands. If so, the process advances to step 4820 where theinput signal to be bandsplit is retrieved, and then advances to step4825 where the bandsplit filter and scaling process, typically similarto process 1178 of FIG. 10E, is executed. Then, whether as the result ofa false return at step 4815 or the bandsplit filter process at step4825, the process advances to step 4830 where the results are saved forband group output processing. The loop then returns to step 4810 forprocessing the next band. Once the last of the bands have beenprocessed, the process advances from step 4810 to a band group outputprocessing loop at step 4835 for 1 to p bands or band groups. The loopincludes a signal combiner step 4840, followed by a soft clip process4845, and then a bandsplit filter and scaling process at step 4850. Theresults then undergo an output conversion step 4855, after which theprocess returns to step 4835 for processing of the next band group ofthe channel. Once the last band group has been processed, the loopreturns and the process advances from step 4835 to step 4860 where allthe output conversion reference values are obtained and combined into achannel reference value which may be broadly thought of as a referencesignal generator. The process then returns to step 4805 where the nextchannel is processed. Once the last channel is processed, the processadvances from step 4805 to exit at step 4865. Not all processing stepsneed to be executed, depending on the implementation, examples of whichare shown in connection with FIGS. 45A-G.

[0401] Turning next to FIG. 49, an exemplary output conversions process4855 is illustrated in flow diagram form. The process, which generallycorresponds to the logic of FIG. 44, starts at step 4900 and advances tostep 4905 where all appropriate inputs are retrieved and combinedtogether. The inputs are typically generated by bandsplit step 4850 foruse in speaker equalization and the combining is typically achieved by,for example, scaling each input and summing together although otherarithmetic functions may be appropriate in particular embodiments. Thesummed inputs are used to generate the output conversion referencesignal (4345A-n, FIG. 43B) which is saved at step 4910 for later use bythe channel reference generator 4860. The process then advances to step4915 where a test is made to determine whether analog outputs areneeded. If so, the process advances to step 4920 where a signalprocessing step is implemented, typically to allow for scaling of thedata word length. The maximum output gain value is then retrieved atstep 4925 and applied to the linear power amplifier in step 4925,followed by a D/A conversion and linear power amplification step 4930.

[0402] After converting and amplifying the analog signal at step 4930,or if no analog signal was needed as determined at step 4915, theprocess advances to step 4935 where a test is made to determine whetherdigital outputs are needed. It should be noted that both analog anddigital outputs may be generated substantially simultaneously, allowingfor maximum flexibility. If digital outputs are needed for theparticular system implementation, the process advances to step 4940 forsignal processing, which typically allows for data word length scaling.The process advances to step 4945 where the output sample is placed intothe output FIFO buffer, and then progresses to step 4950 where a test ismade to determine whether the digital outputs need to be packetized forthe particular implementation. If so, the process advances to step4955,where a test is made to determine whether the next packet is ready.If so, the packet is generated at step 4960 otherwise the cycle isskipped until enough data has been processed into the FIFO to make apacket and test 4955 becomes true Once the packet is sent, or if nopacket is ready, the process advances to step 4970 and exits, which alsooccurs at step 4935 if no digital outputs are required. If a falseresult occurs at test 4950 (i.e., no packets needed), the processadvances to step 4965 where the next sample is output from the FIFO,typically for constant rate devices such as digital tape recorders,after which the process advances to exit step 4970.

[0403] Turning to FIG. 50, the calibrator/annunciator 420 of FIG. 4 canbe better appreciated. The calibrator annunciator performs two generalfunctions: first, it provides calibration signals for several purposes,and second it provides annunciator functions to the user. In particular,the calibrator provides calibration signals for noise compensation loopbalancing, as discussed hereinafter, provides calibration signals forautomatic channel/band balancing of the system, and can download newcalibration signals. The annunciator portion functions to provides voicefeedback to the user including instructions to the user to changesettings. The annunciator can include a variety of techniques from voicesynthesis to voice compression, and can digitally record a message. Thecalibrator/annunciator 420 can perform arbitrary waveform generation,white noise generation, Fourier synthesis, computed noise, and/orAM/FM/PM signals to produce output calibration signal 5010, which isprovided to the signal bus 400B for dissemination through the system.Control Bus 400A may be used to select the annunciator message, selectthe calibration generation method, and record input signals from SignalBus 400B.

[0404] Turning to FIGS. 51 through 64, the noise compensation aspect ofthe overall system of the present invention may better appreciatedbeyond the discussion provided in connection with FIG. 4, above. Ingeneral, the ambient environmental noise portion of total environmentalinput 470 is detected with compensation for the sound signal of thesystem, and is used to increase the system outputs to overcome suchambient noise or, depending on the user's preferences, to allow thesystem output to be reduced to give the ambient noise (such as aconversation) priority over the system outputs. In general, there arethree variations of noise compensators, i.e. closed loop, leakage loop,and open loop. The closed loop is the most robust, while the othervariations trade off increased acoustic restrictions for reducedprocessing requirements.

[0405] In a closed loop noise compensator, the environmental sensors,typically microphones, that detect the environmental signal 470 (fromFIG. 4, above and FIG. 51) is assured of detecting the entire soundsignal produced by the speakers 480. Since all of the sound generated bythe speakers is detected by the microphone, it is possible to calibratethe response of the loop processor 1200 and 1205 (FIG. 11, above andFIG. 51) to a given or plurality of channel reference outputs 4310(FIGS. 11 and 43, above) by “system” or “loop” balancing. For optimalnoise compensation, negative loops are used in the loop processor1200/1205 to ensure stability in any acoustic environment, inparticular, to avoid a gain chase problem that results from a changingacoustic environment due to variations in room acoustics and resonances.

[0406] In a leakage loop noise compensator, the loop processor 1200 and1205 obtains a partial signal 470 from the speakers 480, typically theresult of sound leakage from a headphone or handset, and the remainderof signal 470 from environmental noise. A negative loop 1205 is requiredin the loop processor to compensate for variations in the amount ofsound leakage. Calibration of an open/closed loop noise compensator may,in some embodiments, be avoided by proper design, or by a one timefactory calibration that matches environmental noise levels to theoutput signal levels; e.g. that produced by a headphone's speakers.

[0407] An open loop noise compensator has no acoustic coupling betweenthe speakers 480 and the environmental sensors that detect environmentalinput signal 470, thus no negative loop is needed to obtain stableoperation. Examples of environmental sensors without acoustic couplinginclude speedometers, accelerometers, tachometers, and status indicatorssuch as window up or down. Calibration of an open loop noise compensatorcan be avoided by proper design.

[0408] With the foregoing in mind, the exemplary arrangements shown bythe figures may be better understood. Referring first to FIG. 51, thereis shown therein a generalized noise compensation loop using variouscomponents of the partitioned signal processing system. To simplify theconceptual noise compensation loop example of FIG. 51, a single band,single channel compander and a single environmental sensor (microphone)are shown from which it will be apparent to those skilled in the art,given the other teachings herein, that multiband, multichannelcompander, multiple environmental sensor, systems using open and closedloops simultaneously, and volume control only systems can also berealized. The loop, also referred to as the positive loop, begins withan input signal 2005, typically 0 dB adjusted signal 860, being suppliedto compander 450, the output of which is supplied to volume control 445and to the output signal processor 475, which in turn provides theamplified signal that drives speaker 480 and produces a channelreference out signal 4310.

[0409] The noise extractor 465 is comprised of two parts, loop processor1200 and 1205, and a noise processor 1210. The total environmental input470 supplied to the loop processor 1200 and 1205 is processed with thechannel reference out 4310 signal to provide positive loop outputs 5100supplied to the noise processor 1210. The negative loop processing istypically performed locally in loop processor 1200 and 1205 or inconnection with noise processor 1210 via the use of noise feedback 5105.The noise processor 1210 in turn generates a compander noise floor 5110or volume control noise offset 5115 signal which are supplied to thetransform engine 410. The transform engine 410 in turn controls thefunction of the compander 450 and volume control 445, as a function ofsignals 5110 or 5115 and user interface 405.

[0410] FIGS. 52A,B,C show block diagrams of loop processor 1200,positive/negative loop comparisons 1205, and noise processor 1210. Notall elements shown in the block diagrams are required in everyembodiment. Exemplary implementations and the elements used therein aredescribed in greater detail in connection with FIGS. 53A,E,F and 54A-F.

[0411]FIG. 52A shows the Loop Processor block diagram comprising threemajor sections. The first section processes the environmental signals470 and may include the input adjust blocks 5300A-e, negative loopfeedback 5305A-f, signal conditioning and delay blocks 5310A-k,combiners 5325A-t, and negative loop feedback control block 5302A. Thesecond section processes the reference signals 4310 and may include theinput adjust blocks 5300B-j, negative loop feedback 5305B-g, signalconditioning and delay blocks 5310C-m, combiners 5325B-v, and negativeloop feedback control block 5302B. The third section is the loop balanceprocessor 5360.

[0412] At calibration time, when there is a minimum of environmentalnoise, the acoustic loop balancing processor 5360 receives inputs fromthe environment power estimate bus 5220 and the reference powerestimator bus 5215 and adjusts the environment balance gain 5364A-h,environment fine adjust values 5362A-h, reference balance gain 5364B-j,and reference fine adjust values 5362B-j supplied to the input adjustblocks 5300A-e, B-j and negative loop feedback control blocks 5302A,B,until the environment power estimate bus values are the same as thereference power estimator bus values. Input adjust block 5300 typicallyconsists of one or more multipliers controlled by a balance gain signal5364 and fine adjust signal 5362 to increase or decrease the signalvalues 470 or 4310. An alternate method of fine input adjustment usesthe negative loop feedback control 5302, negative loop feedback 5305 andfine adjust 5362 to adjust the input signal amplitude. Loop balanceprocessor 5360 may also provide environment delay constants 5313A-k tosignal conditioning and delay blocks 5310A-k, and reference delayconstants 5313C-m to signal conditioning and delay blocks 5310C-m, tocompensate for system or acoustic propagation delays. Loop balancing maynot be required for all embodiments; for example, such loop balancingmay not be required for leakage loop or open loop systemimplementations.

[0413] An alternative to loop balance processor 5360 is to achieve userbalancing via the user interface 405 and control bus 400A, connected toinput adjust blocks 5300A-e and 5300B-j (shown in FIG. 52A.) By thisapproach, the user may manually adjust the gain of input adjust block5300 until subjectively satisfactory noise compensation is achieved,with the gain value being sent to the input adjust block 5300 via thecontrol bus 400A. It will be appreciated that this approach could alsobe implemented in combination with the loop balance processor 5360; thetwo approaches need not be mutually exclusive.

[0414] In the first section, which processes the environmental signals,environmental inputs 470 are processed into environment power estimatorbus values 5220 which are supplied to the positive loop 5210A-n andnegative loop 5205A-m comparison blocks of FIG. 52B. Input adjustedsignals from input adjust blocks 5300A-e may be supplied toenvironmental power estimate bus 5220 or supplied to the negative loopfeedback blocks 5305A-f. Negative loop feedback blocks 5305A-f may alsoreceive inputs from the environment power estimator bus 5220, forexample to process outputs from combiners 5325A-t. The negative loopcomparison signals from 5205A-m may be used by negative loop feedbackcontrol 5302A to generate a loop gain value 5347A-f, for negative loopfeedback blocks 5305A-f, which typically consist of a multiplier orother gain element controlled by the gain value, to implement primarynegative loops. It will be appreciated that the negative loopcomparisons counteract the differences between the environmental inputsand the reference signals up to various limits, typically determined bythe particular implementation. Beyond these limits, the positivedifferences are assumed to be due to noise, and the positive loops areused to generate the apparent noise floor. Typically the negative loopshave a faster response rate than the positive loop comparisons.Secondary negative loops, which typically respond slower than theprimary negative loops, can be implemented by using reference powerestimator bus signals 5215 or noise processor bus signals 5240 innegative loop feedback control. Secondary negative loops are typicallyused to limit the amount of compression or prevent severe gain chaseproblems. Signal conditioning and delay blocks 5310A-k may receive inputsignals from the negative loop feedback blocks 5305A-f, input adjustblocks 5300A-e, and combiners 5325A-t via environmental power estimatorbus 5220. Typical signal conditioning involves, for example, bandpass orlowpass filtering, Fourier transforms, and/or decimation to reducedigital processing requirements. The input signals may also be delayedto compensate for acoustic or processing delays, the delay value beingdetermined by design or provided by loop balance processor 5360.Combiners 5325A-t may receive input signals from the signal conditioningand delay blocks 5310A-k, negative loop feedback blocks 5305A-f or inputadjust blocks 5300A-e via environmental power estimator bus 5220. Two ormore of these signals may be combined into a single value, typically bya mixer function or a selection function where the appropriate signal isselected (e.g. the largest value signal). The output of combiners5325A-t are provided to the environmental power estimator bus 5220.

[0415] The functions of the second section are similar to the firstexcept that reference inputs 4310 are processed into reference powerestimator bus values 5215 which are supplied to the positive loopcomparison blocks 5210A-n and negative loop comparison blocks 5205A-m ofFIG. 52B. Input adjusted signals from blocks 5300B-j may be supplied toreference power estimate bus 5215 or supplied to the negative loopfeedback blocks 5305B-g. Negative loop feedback blocks 5305B-g may alsoreceive inputs from the reference power estimator bus 5215, for exampleto process outputs from combiners 5325B-v. The negative loop comparisonsignals from blocks 5205A-m may be used by negative loop feedbackcontrol 5302B to generate a loop gain value 5347B-g for negative loopfeedback blocks 5305B-g, which typically consist of a multiplier orother gain element controlled by the gain value, to implement primarynegative loops. Secondary negative loops, which typically respond slowerthan the primary negative loops, can be implemented by using environmentpower estimator bus signals 5220 or noise processor bus signals 5240 innegative loop feedback control. Signal conditioning and delay blocks5310C-m may receive input signals from the negative loop feedback blocks5305B-g, input adjust blocks 5300B-j, and combiners 5325B-v viareference power estimator bus 5215. Typical signal conditioning involvesbandpass or lowpass filtering, Fourier transforms, and/or decimation toreduce digital processing requirements. The input signals may also bedelayed to compensate for acoustic or processing delays, with the delayvalue being determined by design or provided by loop balance processor5360. Combiners 5325B-v may receive input signals from the signalconditioning and delay blocks 5310C-m, negative loop feedback blocks5305B-g or input adjust blocks 5300B-j via reference power estimator bus5215. Two or more of these signals may be combined into a single value,typically by a mixer function or a selection function where theappropriate signal is selected (e.g. the largest value signal). Theoutputs of combiners 5325B-v are provided to the reference powerestimator bus 5215.

[0416]FIG. 52B shows the positive and negative loop comparisons blockdiagram. Environmental power estimator bus 5220 and reference powerestimator bus 5215 are provided to negative loop compare blocks 5205A-mand positive loop compare blocks 5210A-n. Negative loop compare blocksprovide signals to negative loop outputs bus 5225 for negative loopnulling purposes. Negative loop compare blocks typically includeprocessing of the environment and reference power estimate values by alow pass filter for loop stability and/or decimator to reduce processingrequirements, a subtractor to calculate the difference between thefiltered/decimated reference and environment power estimator bus values,and a difference- to-gain converter and limiter to convert and limit thedifference into a negative loop gain offset value which is provided tonegative loop output bus 5225. The filtered and/or decimated environmentand reference signals may also be provided to the environment powerestimator bus 5220 and reference power estimator bus 5215 for use byother positive and negative loop compare blocks to implement serialfiltering of the power estimation signals. The positive loop compareblocks typically include processing of the environment and referencepower estimate values by a low pass filter for loop stability and/ordecimator to reduce processing requirements, a subtractor to calculatethe difference between the filtered/decimated environment and referencepower estimator bus values, and an absolute value function so that onlypositive values representing a noise floor signal are generated andprovided to positive loop outputs bus 5100. The filtered and/ordecimated environment and reference signals may also be provided to theenvironment power estimator bus 5220 and reference power estimator bus5215 for use by other positive and negative loop compare blocks toimplement serial filtering of the power estimation signals.

[0417]FIG. 52C shows an exemplary embodiment of the noise processor inblock diagram form. The positive loop outputs 5100 and reference powerestimator bus 5215 provide input signals and internal configuration bus640 provides user settings to noise processor bus 5240. For theembodiment shown, there are four processing sections that receive inputsand commands from and provide outputs to the noise processor bus 5240.This allows for the sections to be serially connected in a variety ofways, examples of which are provided in connection with FIGS. 54A-F.Noise processor bus 5240 also provides the section outputs externally,for example as the noise feedback and noise compensation signals 5235.It will be apparent from that discussion that not all sections arerequired for various implementations.

[0418] The first section is the corrections blocks 5227A-n whichcomprises differential error eliminator 5400 and negative loopcorrection 5405. In general, the eliminate differential error function5400 and negative loop error correction 5405 modules compensate forerrors introduced by negative loop nulling, typically with one set ofelements 5400 and 5405 per negative loop. Differential error eliminator5400 and negative loop correction 5405 are discussed in more detail inconnection with FIG. 54A, hereinafter.

[0419] The second section is the sensitivity control which consists ofsensitivity control block 5430A-m and signal combiner block 5230A-m. Thesensitivity control block 5430 allows the user to alter the systemsignal- to-noise ratio by changing the noise floor signals 5110 and5115. The sensitivity control is discussed in more detail in connectionwith FIGS. 54A,B,C. A signal combiner 5230 may be used to combinemultiple inputs from positive loop outputs 5100, corrections block 5227,volume control offset block 5445, and variable attack/release block 2275into one signal provided to a sensitivity control block 5430. Typicalcombiner functions include mixing together two or more inputs orselecting the appropriate input (e.g. maximum value input).

[0420] The third section is the volume control offset which consists ofvolume control offset block 5445A-o and signal combiner block 5230B-o.The volume control offset block 5445 calculates the additional volumecontrol gain required to compensate for environmental noise in volumecontrol only systems. The volume control offset block 5445 is discussedin more detail in connection with FIGS. 54B,C. A signal combiner 5230may be used to combine multiple inputs from positive loop outputs 5100,corrections block 5227, sensitivity control block 5430, and variableattack/release block 2275 into one signal provided to a volume controloffset block 5445. Typical combiner functions include mixing togethertwo or more inputs or selecting the appropriate input (e.g. maximumvalue input).

[0421] The fourth section is the variable attack/release which consistsof variable attack/release block 2275A-p and signal combiner block5230C-p. The variable attack/release function produces a long durationnoise floor. The variable attack/release function is discussed in moredetail in connection with FIGS. 22, 24A-G, 54A, 55A-C, 56, and 57A,B. Asignal combiner 5230 may be used to combine multiple inputs frompositive loop outputs 5100, corrections block 5227, sensitivity controlblock 5430, and volume control offset block 5445 into one signalprovided to a variable attack/release block 2275. Typical combinerfunctions include mixing together two or more inputs or selecting theappropriate input (e.g. maximum value input).

[0422] Turning to FIG. 53A, there is shown therein an example of how aloop processor may be implemented using a loop input processor 1200, asingle negative loop comparison 5205 and a single positive loopcomparison 5210. The control bus 400A provides control signals to anacoustic loop balancing processor 5360, which also receives a pluralityof other inputs which will be discussed hereinafter. The acoustic loopbalancing processor 5360 generates a microphone balance gain signal5364A and microphone fine adjust signal 5362, which are determinedduring the calibration of the noise loop and represent the amount ofgain necessary to balance the total environment input 470 with thechannel reference out signal 4310, as will be discussed hereinafter inconnection with FIGS. 62 and 63.

[0423] The environmental input 470 is supplied by a microphone andamplified by input level adjuster 5300A using the microphone balancegain signal 5364A. In this example, adjuster block 5300A is a coarsegain adjustment, typically included in commercially available codecs.The signal from adjuster block 5300A is then supplied to the negativeloop feedback 5305 which is provided with an environmental negative loopgain value 5347 by negative loop feedback control 5302. Processor 5360supplies microphone fine adjust signal 5362 (which in conjunction withmicrophone balance gain 5364A provide accurate microphone gainadjustment), and a negative loop gain offset signal (negative loopoutputs bus) 5225 to feedback control block 5302 to effect negative loopnulling. Negative loop feedback 5305 typically consists of multipliersor other gain elements. The signal from 5305 undergoes signalconditioning 5310A, typically lowpass or bandpass filtering, resultingin fast environment power estimator signal (environment power estimatorbus signal) 5220, which is part of the negative loop comparison 5205.The signal 5220 is supplied as the minus signal to a subtractor 5315which is the negative loop comparison element. The signal 5520 is alsosupplied to another signal conditioning block 5310B, typicallyadditional lowpass filtering and/or decimation to reduce processingrequirements, as part of positive loop comparison 5210. The output ofthe second signal conditioning block 5310B is slow environment powerestimator signal 5335 of environment power estimator bus 5220, and issupplied back to the acoustic loop balancing processor 5360 for loopbalancing as well as being supplied to another subtractor 5350 which isthe positive loop comparison element. The signal 5335 is also suppliedas an apparent noise floor output for leakage and open loop systems fromFIG. 53A as shown at the bottom of the Figure.

[0424] The channel reference out 4310, which may be multiple signals,provides the other side of the comparison developed by FIG. 53A. Thesignal(s) 4310 are supplied from the control bus 400A to input adjustblocks 5300B-k, which each receive a control signal in the form ofnegative loop bias signal 5365 and reference balance gain signal 5364B-kto determine the amount of input gain. The signal 5365, which isdetermined by design, is also provided to the processor 5360. The biassignal 5365 allows for positive gain that is typically required for thenegative loop. The output of input adjust blocks 5300B-k is provided tofurther signal conditioning blocks 5310C-k, the output of which formsthe fast channel reference power estimates signals 5312 of referencepower estimator bus 5215 which is supplied to combiner block 5325 toconvert the multiple inputs into a single output, typically by use of amultiple input mixer with input scaling. The output of the block 5325forms a fast system reference power estimator signal on reference powerestimator bus 5215, which is supplied to the positive side of thenegative loop comparison subtractor 5315, a further signal conditioningblock 5310D which forms part of the positive loop comparisons, andfinally is supplied as one input to the negative loop comparisonΔ-to-Gain converter 5345. The output of the signal conditioning block5310D, typically additional lowpass filtering and/or decimation toreduce processing requirements, forms a slow system reference powerestimator signal 5340 on reference power estimator bus 5215 and issupplied as the negative side input to the positive loop comparisonsubtractor 5350, is also fed back to the processor 5360 for loopbalancing and forms an output from the function of FIG. 53A. The outputof the subtractor 5350 is typically zero if there is no environmentalnoise or positive if there is noise so negative values are removed byelement 5355, typically an analog rectifying diode or absolute valuecalculation, to form an apparent noise floor for closed loops signal5395. Finally, the output of the Δ-to-Gain converter 5345 forms anegative loop gain offset signal on negative loop outputs bus 5225 andis provided to the negative loop feedback control block 5302. The block5302 also receives a microphone fine adjust signal 5362 from theprocessor 5360.

[0425] At calibration time, when there is a minimum of environmentalnoise, the acoustic loop balancing processor 5360 determines the properclosure coefficient constants (microphone balance gain 5364A, microphonefine adjust 5362, and reference balance gains 5364B-k) necessary tobalance the signals from the microphone(s) 470 with the channelreference(s) out 4310 signals. This processor can use any signals fromthe environment and reference power estimator busses 5215 and 5220 indetermining these constants. In the example of FIG. 53A, the slowenvironment and slow system reference power estimates (5335, 5340) ofthe environment and reference power estimator busses are used forbalancing.

[0426] The negative loop comparison 5205 computes the difference 5315,and provides this value with the fast system reference power estimator5215, to the A to gain converter 5345. The A to Gain converter 5345generates a limited negative loop gain offset 5225 that is provided tonegative loop feedback control 5302 to null out any differences betweenthe fast environment power estimator signal 5220 and the fast systemreference power estimator 5215 (which is composed of a plurality ofchannel reference out 4310). The negative loop comparisons 5205optionally perform additional signal processing on the fast environmentpower estimator and fast system reference power estimator signals, thesignal processing typically being lowpass filtering.

[0427] The positive loop comparison 5210 includes performing signalconditioning 5310B and D on the fast environment power estimator 5220and the fast system reference power estimator signal 5215 to produce theslow microphone power estimator signal 5335 and slow system referencepower estimator signal 5340 which are included in the environment andreference power estimator busses 5220 and 5215. The positive loopcomparisons 5210 computes the difference 5350 of these signals, andselects positive values only at 5355, resulting in the apparent noisefloor for closed loops 5395. If a leakage or open loop configuration isto be used, then the apparent noise floor for leakage/open loops 5335 isobtained from the slow microphone power estimator 5335 since the systemreference need not be subtracted out. Signals 5335 and 5395 are providedto the positive loop outputs bus 5100.

[0428] It will be appreciated from the foregoing that many othervariations are possible. For example, the input to the positive loopsignal conditioning can come from the negative loop input or output ofthe negative loop signal conditioning for serial or parallel loop use.Also the negative loop may be included in the reference signalprocessing loop.

[0429] Referring next to FIGS. 53B and C, there are shown thereindetailed implementation examples of the negative loop feedback controlblock 5302 and negative loop feedback block 5305. FIG. 53B shows the“sum of offsets” implementation where all inputs contributing to thenegative loop feedback 5305 multiplier gain control are summed togetherto produce negative loop gain signal 5347. FIG. 53C shows the “productchain” where each input contributing to the negative loop is provided toa multiplier in the negative loop feedback block 5305. Hybridimplementations where a combination of the “sum of offsets” and “productchain” implementations are also possible.

[0430] Referring next to FIG. 53D, there is shown therein a detailedimplementation example of the negative loop comparison 5315 andΔ-to-gain converter 5345. The negative loop nulls the difference betweenthe fast environment power estimator 5220 and fast system referencepower estimator 5215 signals over a limited range (negative loopwidth=+/−N) to compensate for minor variations in the environmentalsignal caused by changes in room acoustics and room resonances. Thisloop is implemented as a standard differential negative loop.

[0431] Loop gain is required to optimize the nulling effect. It isimportant to calculate the optimal loop gain value. If the loop gain isexcessive, the negative loop oscillates. If the loop gain is too small,it results in a larger differential error, making the negative loop lesseffective. Multiplying the reference-microphone difference 5315 by thenegative loop gain in multiplier 5394 provides the A loop signal.

[0432] A negative loop limit value is generated to establish the limitedrange over which the negative loop operates. This limit may becalculated by multiplying the fast system reference power estimator 5215with a negative-loop width signal (normally set at a constant valuebased on the negative loop width “N”) in multiplier 5396. If thenegative loop width signal is varied, it acts as a variable negativeloop control, such as when supplied by noise feedback 5105, which may beused as a secondary negative loop along with a primary fixed width loop.

[0433] The negative loop gain offset calculator 5398 of A to gainconverter 5345 computes the ratio of the A loop 5394 to the negativeloop limit 5396 and uses that ratio to compute the percentage of maximumor minimum negative loop gain offset 5225 required to null the loop.This percentage gain can be discretely quantified. In the extreme caseof 100% quantization, the negative loop gain offset 5225 is the maximumor minimum if it exceeds the negative loop limit or zero. Withsubsequent low pass filtering, this effectively implements a pulse widthmodulated negative loop.

[0434] In situations where there are long acoustic propagation delaysfrom speakers 480 to environmental input microphones 470, such as in astadium, acoustic delay compensation may be used to increase thestability of the system as shown in FIGS. 53E and 53F. In both figures,acoustic loop balancing processor 5360 supplies reference delayconstants 5313, which indicates the length of the delay, to delaycompensation elements of signal conditioning and delay compensation5310, typically composed of acoustic delay lines or digital FIFObuffers. Acoustic loop balancing processor 5360 may generate delayconstants during loop balancing or they may be determined by design.FIG. 53E shows an example of a robust implementation where each channelreference out 4310 has a delay compensation element 5310D-k. This isuseful in situations where the distance (and acoustic delay) from thevarious speakers to the environmental input microphones varies over awide range and individual compensation of each reference provides thebest results. For situations where the distance (and acoustic delay)from the various speakers to the environmental input microphones varieslittle, a reduction in delay compensation elements 5310 and processingrequirements can be obtained by applying the delay compensation 5310Donly to the system reference signal generated by combiner block 5325 asshown in FIG. 53F.

[0435] Turning next to FIG. 54A, an exemplary embodiment of the noiseprocessor 1210 is shown when used with a compander. In general, thelogic of FIG. 54A is intended to permit the noise processor 1210 to takethe apparent noise floor signals, which may originate as either signals5335 or 5395 provided by the positive loop output bus 5100, and correctthem into a compander noise floor 5110, thus enabling more accurate andeffective noise compensation. These noise floors can be made to appearlarger or smaller than the actual value by a noise sensitivity control5440 and sensitivity control block 5430. In the example of FIG. 54A, theapparent noise floor signals are provided to block 5400 which functionsto eliminate any differential error introduced by negative loops. Theoutput of the block 5400 is a differential corrected noise floor signal5402, which is supplied to a negative loop error correction block 5405.A correction convergence factor 5407, determined by design and discussedlater in connection with FIGS. 54G and 54H, as well as the slow systemreference power estimator signal 5340 also serve as inputs to thenegative loop error correction block 5405. The output of the correctionblock 5405 is a corrected noise floor signal5417, and is supplied to thevariable attack and release block 2275, the output of which is a longduration noise floor 5425. The long duration noise floor signal 5425 aswell as noise sensitivity control signals 5440 from the control bus 400Aserve as inputs to a sensitivity control adjuster 5430, which outputsthe compander noise floor signal 5110.

[0436] The eliminate differential error function 5400 eliminates theerror due to the differential nature of the primary negative feedbackloop. Increasing the negative loop gain minimizes this error. Toeliminate this error, typically the maximum calculated error issubtracted from the apparent noise floor. The negative loop errorcorrection 5405 (discussed in greater detail in connection with FIG.54G) eliminates the error in the differential corrected noise floor 5402caused by the width of the negative loop in the negative loopcomparisons 5205. The decimator/low-pass filter of 5405 can be used toreduce the amount of subsequent computation and provide low-passfiltering. The variable attack and release 2275 allows ignoring shortduration, transient noises, while responding to longer duration noises.It also enables faster response to large, long duration noises, than tolower amplitude long duration noises. The sensitivity control adjuster5430 allows the noise sensitivity control 5440 to alter the systemsignal to noise ratio by changing the compander noise floor 5110 signal.The sensitivity control adjuster 5430 may convert the long durationnoise floor 5425, for example, into a logarithmic value to simplifysubsequent processing in which case the sensitivity control 5440 can bea logarithmic value and added or subtracted from the long duration noisefloor to increase or decrease the system signal to noise ratio.

[0437] Turning next to FIG. 54B, an example is shown of how a noiseprocessor 1210 could be implemented to control a volume control. Likeelements have been assigned like reference numerals and will not bedescribed further except as necessary for the example. The apparentnoise floor 5335 or 5395 is transformed into a corrected noise floor5410 by the differential error eliminator block 5400 together with thenegative loop error correction 5405. The corrected noise floor signal5410 is supplied to the sensitivity control adjuster block 5430, in thisexample typically a multiplier with the noise sensitivity control 5440value being a linear quantity. The output of the sensitivity controladjuster block 5430 supplies the positive input to a subtractor ofvolume control offset calculator 5445, with the slow reference powerestimator 5340 supplying the negative input. The output of thesubtractor is limited to positive values, typically by use of a diode orabsolute value function, for use as a volume control offset signal 5419and may be supplied to a decimator/low pass filter block. This output,in turn, is supplied to a variable attack and release block 2275 whichoutputs volume control noise offset signal 5115, typically used toincrease the volume control level to compensate for ambient noise.Optional linear to log conversion may be performed in block 2275 whichmay be used to simplify subsequent processing.

[0438] Turning next to FIG. 54C, an example is shown of how minimalnoise processor 1210 with sensitivity control could be implemented tocontrol a volume control. Here the slow environment power estimate 5335(an uncorrected positive loop output 5100) is multiplied with noisesensitivity control 5440 to implement the sensitivity control, whichthen has the slow system reference power estimate 5340 subtracted fromit to produce the volume control offset 5115. The volume control offset5115 is typically limited to positive values, for example by use of adiode or absolute value function.

[0439] FIGS. 54D,E,F show examples of how the elements described inconnection with FIG. 52C can be configured to accommodate multiplepositive loop inputs and compander noise floor outputs. Like elementshave been assigned like reference numerals and will not be describedfurther except as necessary for the example. FIG. 54D shows an exampleof how multiple positive loop outputs 5100 can be processed bycorrections blocks 5227A-n, Variable attack/release blocks 2275A-n, andsensitivity controls blocks 5430A-n to provide “n” independent compandernoise floor signals 5110 A-n. FIG. 54E shows an example of how threepositive loops can be used to produce a single compander noise floor5100. Each positive loop output 5100 is corrected (blocks 5227A,B,C) andcombined in signal combiner 5230 to provide a single signal to variableattack/release and sensitivity control blocks 2275 and 5430 to producecompander noise floor signal 5110. FIG. 54F shows an example of how asingle positive loop output can be used to control two companders withdifferent attack/release and sensitivity settings.

[0440] Turning next to FIG. 54G, an exemplary arrangement of a negativeloop error correction 5405 may be better appreciated. In general, theeliminate differential error function 5400 and negative loop errorcorrection 5405 modules compensate for errors introduced by negativeloop nulling, typically with one set of 5400 and 5405 per negative loop.The negative loop error correction 5405, shown in FIG. 54G, corrects foran underestimation of the actual noise floor due to the negative loopwidth. Environmental noise is detected when the microphone signal isgreater than the reference signal, but due to the negative loop nulling,the noise is not detected until the microphone signal is greater thanthe reference signal plus the negative loop width “N.” This results inan underestimation of the noise floor.

[0441] The amount of underestimation can be calculated and corrected, anexemplary arrangement of which is discussed in connection with FIGS. 54Gand 54H. In FIG. 54G, the slow system reference power estimator 5340 ismultiplied at multiplier 5460 by a negative loop width correction value,typically a linear constant, to produce the underestimation correctionlimit value 5462. Alternatively, a logarithmic negative loop widthcorrection value may be used after conversion to a linear value bymodule 5455. For implementation of an alternative sensitivity control, avariable negative-loop width correction value may be used to over orunder correct the apparent or differentially corrected noise floor. Thepartially corrected noise floor signal 5452 is generated by multiplyingthe uncorrected noise floor, typically the differential corrected noisefloor 5402, by a correction convergence factor 5407 in multiplier 5450.The correction convergence factor determines how quickly the correctednoise floor 5410 becomes fully corrected. The partially corrected noisefloor signal 5452 and the correction limit 5462 both serve as inputs toa select minimum value function 5465, which is added in adder 5470 tothe differential corrected noise floor signal 5402. The output of theadder 5470 is corrected noise floor signal 5410.

[0442] Referring next to FIG. 54H, a graphical analysis of the FIG. 54Gnegative loop error correction example is discussed. Stated otherwise,the apparent noise floor can be seen to be the difference between themicrophone power estimate and the system reference power estimate. Thehorizontal axis represents the difference between the microphone powerestimate and the system reference power estimate and the vertical axisis the indicated noise floor value. Without any negative loops, themicrophone power estimate and the system reference power estimatedifference is the actual noise floor value as indicated by the “actualnoise floor” line on the graph. Due to the use of negative loops, themicrophone power estimate and the system reference power estimatedifference will underestimate the noise floor as represented by the“apparent or differential corrected noise floor” line on the graph whichmay be corrected to the actual noise floor value by various methodsdescribed below.

[0443] Always adding correction limit 5462 to the noise floor results ina minimum noise floor always being detected. Adding the correction uponnoise floor detection causes an undesirable discontinuity, as indicatedby the infinite correction gain in FIG. 54H.

[0444] To obtain a more gradual correction, the apparent noise floor5335 or 5395 can be multiplied by the correction convergence factor 5407to generate a partially corrected noise floor 5452, which is added tothe apparent noise floor 5335 or 5395 to produce corrected noise floor5410 until the partially corrected noise floor 5452 value exceeds thecorrection limit 5462 at which point the correction limit is added. Theresulting corrected noise floor value 5410 is indicated in the graph asthe two bold lines labeled “partially corrected noise” and “fullycorrected noise”. The correction convergence factor controls the slopeof the “partially corrected noise” line. The select minimum value 5465selects the partially corrected noise floor 5452 value or the correctionlimit 5462, producing the two lines.

[0445] In the example shown in FIG. 54H, the correction limit can becomputed as shown in the following example using a negative loopwidth=+/−2 dB.

[0446] The start of noise floor detection is when:

[0447] microphone−reference=2 dB

[0448] Rewriting this equation provides:

[0449] microphone=2 dB+reference.

[0450] By substitution, the correction limit shown in the graph may becomputed as:

[0451] correction limit=(2 dB+reference)−reference

[0452] correction limit=(10**(2 dB/20)) * reference−reference

[0453] correction limit=1.26* reference−reference

[0454] correction limit=0.26* reference

[0455] In general:

[0456] correction limit=(10**(N dB/20)−1) * reference

[0457] Note that an alternative noise sensitivity control method to thatpreviously described in connection with module 5430 may be implementedby increasing the correction limit above that calculated which resultsin an overestimation of the noise floor and increases noise sensitivitywhile decreasing the limit results in an underestimation of the noisefloor which decreases noise sensitivity.

[0458] Referring next to FIGS. 55A and 55B, various examples of thevariable attack and release portion of the noise compensation functioncan be better appreciated. In a noise processor 1210, the variableattack/release function 2275 may be configured to provide minimalresponse to short duration periods of noise (such as door slams, shortburst of speech, and transient sounds), while also providing rapidresponse to long duration noises such as machinery or road noise.

[0459] For at least some embodiments, the characteristics of a usefulresponse to changes in the noise floor signal are as follows: Whenchanges in the noise signal occur, the initial response should be todelay, typically by a delay timer or by integrating the signal. If thechange in the noise signal is longer lasting, then there should be aquick convergence on the proper noise floor, preferably with increasingexponential response for attack, and decreasing exponential response forrelease.

[0460] This response is often desirable because the ear responds tosound in a non-linear manner, so these exponential responses soundlinear to the ear. Once the response has converged on the noise signal,signal distortion is minimized by having a slow response, such asobtained using a conventional low-pass filter. In addition, it isdesirable to have an asymmetric response, with a slower attack andfaster release.

[0461]FIGS. 55A and 55B illustrates the differences between the responseto a change in the noise floor by a prior-art low pass filter and thefilter response provided by the noise compensator's variableattack/release processor. It is not possible to obtain the desiredsignal response using a prior-art low pass filter due to the decreasingexponential attack response that causes it to respond too quickly to theinitial change in the noise floor. Increasing the prior-art low passfilter delay causes an increase in closure error on the actual noisefloor due to the decreasing exponential response. In contrast, thepreferred attack/release processor uses an increasing exponential signalto eliminate closure error.

[0462] Turning to the next figure, FIG. 55C is a block diagram showingan exemplary embodiment of the variable attack/release function 2275 ina noise processor 1210. A math processor 2405 is used to dynamicallycalculate the positive loop differential signal 5620 as a function ofthe external inputs 2400, typically a signal from noise processor bus5240 such as the processed corrected noise floor 5417 or volume controloffset 5419, together with feedback bus values 2415, typically the longduration noise floor 5425. The math processor 2405 also generates adifferential polarity change signal 5625 whenever the polaritydifference between the processed corrected noise floor 5417 and longduration noise floor 5425 occurs (i.e. between the external inputs andfeedback bus values).

[0463] The signals 5620 and 5625, along with internal configurationsignals 640, are supplied to a segment parameter selector 2425, whichgenerates the final filter coefficients 2440, in this example KI 5635,KF 5640, Accelerate Limit 5645, that are supplied to the trackingadjusting filter 2427 (shown in greater detail in connection with FIG.57A), which produces the desired filter response in the form of the longduration noise floor 5425 which may be on noise processor bus 5240. TheJK integrate signal 5725 and long duration noise floor signal 5425 arefed back to the segment parameter selector 2425 and math processor 2405via feedback bus 2415.

[0464] Referring next to FIG. 56, an exemplary embodiment of the mathprocessor 2405 and segment parameter selector 2425 of the variableattack/release function 2275 for a noise processor 1210 enables forminga composite of different signal responses to provide the desired signalresponse shown in FIGS. 55A and 55B. In particular, this embodiment usesa delay integrator, convergence, and low pass filters. In particular,the functions of the math processor 2405 (shown by dashed lines) areimplemented by subtractor 5600 operating on the external input 2400, inthis example a signal from noise processor bus 5240 such as theprocessed corrected noise floor signal 5417, as the positive signal, anda feedback bus signal 2415, the current long duration noise floor signal5425, as the negative input. This provides a positive loop differential5620 that can be either positive or negative, and is supplied to thedifferential polarity change state machine 5605 that indicates when thissignal changes polarity.

[0465] The functions of segment parameter selector 2425 (also shown bydashed lines) are implemented by segment selector 5610 and lookup table5615. The segment selector uses positive loop differential 5620,differential polarity change 5625, JK integrate 5725, slow responselimit 5650, and delay time limit 5655, to select a particular segment(slow response, converge, or delay). User select 5660 and 5665 can beused to provide user selectable different segment responses, for exampleto allow the user to select different delay times, short delays forsporting events and longer ones for more typical use. The lookup table5615 produces the final filter coefficients 2440 (KI 5635, KF 5640, andacceleration limit 5645) as selected by these inputs.

[0466] Turning next to FIGS. 57A and 57B, FIG. 57A shows in blockdiagram form a tracking adjusting filter 2427 as might be used in anexemplary embodiment of a noise compensator variable attack/releasefunction 2275 for a noise processor 1210. FIG. 57B shows exemplarytracking adjusting noise filter signals as might be generated by thevarious configurations of the tracking adjusting filter of the attackand release module of FIG. 57A. The arrangement of FIG. 57A enablesforming a composite of different signal responses to provide the desiredlong duration noise floor 5425 response, in this case by using a delayintegrator, convergence, and low pass filters. These responses arecontrolled by the final filter coefficients 2440 (acceleration limitsignal 5645, KI signal 5635, and KF signal 5640) supplied to it, and maybe modified as desired.

[0467] The delay integrator is implemented when the KF signal 5640equals one causing multiplier 5700 to pass unaltered the value of JKintegrate 5725 to adder 5710. The other input to adder 5710 is the inputstep size generated by multiplying KI 5635 with positive loopdifferential 5620 at multiplier 5705. Typically KI value 5635 is between0 and 1 with larger values of KI accumulating faster resulting inshorter delays. Delay integrator mode is exited when the value of JKintegrate 5725 exceeds the delay time limit 5655 as determined bysegment selector 5610. The acceleration governor, composed of multiplier5700, subtractor 5715, adder 5720, and minimum input difference selector5717, is not required in delay integration mode. The Accelerationgovernor may be disabled by use of a larger acceleration limit value5645 and non-zero positive loop average 5740 multiplied by 5750 whichwill result in an input to minimum input difference selector that isalways larger than the input difference generated by subtractor 5715 sothat the subtractor value is always selected. Adder 5720 then adds backthe JK integrate value that was subtracted at subtractor 5715 andsupplies the input value to JK integrate 5725. Alternatively, theoutputs of segment selector 5610 can be used to disable the accelerationgovernor for modes that do not require the function.

[0468] The operation of a low pass filter is similar to the previouslydescribed delay integrator except that the KF value 5640=1−KI 5635,where KI is between 0 and 1. Since KF is less than 1, multiplier 5700passes a percentage of the value of the JK integrate register 5725,resulting in a steady state condition when KF * JK integrate=KI *positive loop differential. The acceleration governor is also notrequired in low pass filter mode and may be disabled by the methodspreviously described for the delay integrator.

[0469] The function of an accelerating integrator or converge areobtained when the value of KF>1. In this situation, input differencestep size 5715 begins small, and starts to become larger. If notlimited, the step size will become so large that over and undershoots ofthe long duration noise floor 5425 will result.

[0470] Acceleration Limit 5645 limits the rate of acceleration bydetermining the maximum step size to be added to the JK integrate value5725 at adder 5720 by multiplying the positive loop average 5740 by theacceleration limit 5645 at multiplier 5750. The input difference stepsize is calculated at subtractor 5715 where the difference between thenext JK integrate value calculated at adder 5710 and the current JKintegrate value is calculated. The minimum of the two input values isselected by minimum input difference selector 5717 and the next JKintegrate value is restored by adding back the current JK integratevalue at adder 5720 which is then supplied as the input to JK integrate5725. Use of the acceleration governor results in the desiredlogarithmic convergence response without over or undershoots of thenoise floor that can distort the final output signal.

[0471] The JK integrate value 5725 can be a positive or negative valuewhile the long duration noise floor value 5425 is typically a positiveonly value. The JK hold value 5745 is used as a positive bias value toconvert the bipolar JK integrate value 5725 into a positive only valueat adder 5735. The JK integrate register 5725 stores the results of eachstep computation, but is reset whenever a differential polarity change5625 occurs, along with JK hold block 5745. JK hold block 5745 storesthe current long duration noise floor 5425, and when combined by adder5735 with the value of JK integrate register 5725, results in thepositive loop average 5740, or long duration noise floor 5425.

[0472] Turning next to FIG. 57B, there is shown therein a graphillustrating the typical operation of the tracking adjusting filter inFIG. 57A. The three different types of response generated by thetracking adjusting filter 2427 are identified in the waveforms, i.e.delay response (integrator), converge (accelerating integrator), andslow response (low pass filter).

[0473] JK integrate register 5725 acts over positive and negative rangesin response to the positive loop differential 5620 and provides asymmetrical signal response. JK integrate register 5725 is reset to zeroduring transitions between attack and release of the noise-input signalin order to obtain the desired symmetrical response.

[0474] JK hold register 5745 contains a positive bias that when added tothe JK integrate 5725 produces a positive only output value suitable fornoise compensation. The JK hold register 5745 only changes duringtransitions between attack and release by the noise floor, causing thelast long duration noise floor 5425 to be held as a bias signal untilthe next transition. The resulting long duration noise floor 5425 isstored in the positive loop average 5740 register.

[0475] Having described the logic by which the noise extractor 465 andits various elements are implemented, the process of operation for thenoise extractor can be better appreciated by FIG. 58 et seq. Referringfirst to FIG. 58, the overall operation of a generalized form of noiseextractor function as shown at step 710 of FIG. 7, steps 1220, 1235,1245 of FIG. 12, and in FIGS. 11,51-57B, may be better appreciated. Theprocess starts at step 5800, after which the process advances to step5805 where the environmental sensor processing is performed followed bystep 5810 where the reference signal processing is performed. Steps 5805and 5810 are equivalent to step 1220 in FIG. 12 and loop processor 1200in FIG. 11. The process then advances to step 5815 where positive andnegative loop comparisons are performed followed by step 5820 where thenoise floor is determined and the process exits at step 5825. Step 5815is equivalent to step 1235 in FIG. 12 and block 1205 in FIG. 11 and step5820 is equivalent to step 1245 in FIG. 12 and noise processor 1210 inFIG. 11. An alternative processing flow may have one or more stepsoperating in parallel.

[0476] Referring next to FIGS. 59A,B, the processing of theenvironmental sensors step 5805 of FIG. 58 can be better understood. Theprocess starts at step 5900, after which the process advances to step5902 where a loop is begun with the number of iterations of the loopbeing defined by how many environmental sensors are processed. Thenumber of environmental sensors can vary over a wide range, and isidentified here as simply 1 to e. When the process loop begins at step5904, an environmental sensor signal is acquired, the input adjustergain value for that sensor is obtained and the sensor signal adjusted.The loop advances to step 5906, where the adjusted sensor output issaved for later use, after which the process loops back to step 5902.Loop 5902 is equivalent to input adjust blocks 5300A-e of FIG. 52A.

[0477] Once the appropriate number of loops have been completed at step5902, the process advances to step 5908 where all the environmentnegative loops are processed. The number of environmental negative loopscan vary over a wide range, and is identified here as simply 0 to f,zero being used if there are no negative loops. When the process loopbegins at step 5910, bias values and primary and any secondary negativeloop inputs appropriate for this negative loop are obtained and thenegative loop gain value calculated. The process advances to step 5912where the negative loop gain value is applied to the appropriateadjusted environment sensor output from loop 5902 or combiner outputfrom loop 5924 to execute negative loop feedback. The process thenadvances to step 5914 where the results of step 5912 are saved, afterwhich the process loops back to step 5908. Loop 5908 is equivalent tonegative loop feedback control 5302A and negative loop feedback blocks5305A-f of FIG. 52A.

[0478] Once the appropriate number of loops have been completed at step5908, the process advances to step 5916 where signal conditioning anddelay is applied. The number of signal conditioning and delay loops canvary over a wide range, and is identified here as simply 0 to k, zerobeing used if there are no signal conditioning and delay loops. When theprocess loop begins at step 5918, delay values and either theappropriate environment adjusted sensor output from loop 5902, theenvironmental feedback processed output from loop 5908, or combineroutput from loop 5924 are obtained as input values. The process advancesto step 5920 where the signal conditioning and any delay compensation isapplied to the inputs. The process then advances to step 5922 where theresults of step 5920 are saved, after which the process loops back tostep 5916. Loop 5916 is equivalent to signal conditioning and delayblocks 5310A-k of FIG. 52A.

[0479] Once the appropriate number of loops have been completed at step5916, the process advances to step 5923 of FIG. 59B where combining anyof the previously computed output values into a single outputs isprocessed. The number of combiner loops can vary over a wide range, andis identified here as simply 0 to c, zero being used if there are nocombiners. The process loop begins at step 5924 which is a loop tocombine 2 to n of the previously computed output values to a singlevalue. When the process loop begins at step 5926, the appropriateenvironment adjusted sensor output from loop 5902, environmentalfeedback processed output from loop 5908, or signal conditioned anddelayed output from loop 5916 is obtained as an input value. The processadvances to step 5928 where the input value from step 5926 is applied tothe combiner algorithm, after which the process loops back to step 5924.Loop 5924 is equivalent to the combiner blocks 5325A-t of FIG. 52A.

[0480] Once the appropriate number of loops have been completed at step5924, a test is performed at step 5929 to see if the combined outputresult is processed in this pass or saved and used in the next pass byloops 5908 and 5916. If the combined output is to be negative loopprocessed and signal conditioned and delayed in this pass, the processadvances to step 5930 else the process loops back to step 5923. Thedecision to process in the current pass or wait for the next isdetermined by design. At step 5930, bias values and primary and anysecondary negative loop inputs appropriate for this negative loop areobtained and the negative loop gain value calculated. The processadvances to step 5932 where the negative loop gain value is applied tothe combined output from loop 5924 to execute negative loop feedback.The process then advances to step 5934 where the results of step 5932are saved, then to step 5936 where any signal conditioning and delaycompensation is applied and saved in step 5938. The process then loopsback to 5923 and exits at step 5940 once the appropriate number of loopshave been completed at step 5923. Steps 5930, 5932, and 5936 areequivalent to blocks 5302A, 5305A-f, and 5310A-k of FIG. 52A.

[0481] Referring next to FIGS. 59C,D, the processing of the referencesignals step 5810 of FIG. 58 can be better understood. The processstarts at step 5950, after which the process advances to step 5952 wherea loop is begun with the number of iterations of the loop being definedby how many channel references are processed. The number of channelreferences can vary over a wide range, and is identified here as simply1 to j. When the process loop begins at step 5954, a channel referencesignal is acquired, the input adjuster gain value for that referenceobtained and the channel reference adjusted. The loop advances to step5956, where the adjusted reference output is saved for later use, afterwhich the process loops back to step 5952. Loop 5952 is equivalent toinput adjust blocks 5300B-j of FIG. 52A.

[0482] Once the appropriate number of loops have been completed at step5952, the process advances to step 5958 where all the reference negativeloops are processed. The number of reference negative loops can varyover a wide range, and is identified here as simply 0 to g, zero beingused if there are no negative loops. When the process loop begins atstep 5960, bias values and primary and any secondary negative loopinputs appropriate for this negative loop are obtained and the negativeloop gain value calculated. The process advances to step 5962 where thenegative loop gain value is applied to the appropriate adjustedreference output from loop 5952 or combiner output from loop 5974 toexecute negative loop feedback. The process then advances to step 5964where the results of step 5962 are saved, after which the process loopsback to step 5958. Loop 5958 is equivalent to negative loop feedbackcontrol 5302B and negative loop feedback blocks 5305B-g of FIG. 52A.

[0483] Once the appropriate number of loops have been completed at step5958, the process advances to step 5966 where signal conditioning anddelay is applied. The number of signal conditioning and delay loops canvary over a wide range, and is identified here as simply 0 to m, zerobeing used if there are no signal conditioning and delay loops. When theprocess loop begins at step 5968, delay values and either theappropriate channel reference adjusted output from loop 5952, referencefeedback processed output from loop 5958 or combiner output from loop5974 are obtained as input values. The process advances to step 5970where the signal conditioning and any delay compensation are applied tothe inputs. The process then advances to step 5972 where the results ofstep 5970 are saved, after which the process loops back to step 5966.Loop 5966 is equivalent to signal conditioning and delay blocks 5310C-mof FIG. 52A.

[0484] Once the appropriate number of loops have been completed at step5966, the process advances to step 5973 of FIG. 59D where combining anyof the previously computed output values into single outputs isprocessed. The number of combiner loops can vary over a wide range, andidentified here as simply 0 to v, zero being used if there are nocombiners. The process loop begins at step 5974 which is a loop tocombine 2 to p of the previously computed output values to a singlevalue. When the process loop begins at step 5976, the appropriatechannel reference adjusted output from loop 5952, reference feedbackprocessed output from loop 5958, or signal conditioned and delayedoutput from loop 5966 is obtained as an input value. The processadvances to step 5978 where the input value from step 5976 is applied tothe combiner algorithm, after which the process loops back to step 5974.Loop 5974 is equivalent to the combiner block 5325B of FIG. 52A.

[0485] Once the appropriate number of loops have been completed at step5974, a test is performed at step 5929 to see if the combined outputresult is processed in this pass or saved and used in the next pass byloops 5958 and 5966. If the combined output is to be negative loopprocessed and signal conditioned and delayed in this pass, the processadvances to step 5980 else the process loops back to step 5973. Thedecision to process in the current pass or wait for the next isdetermined by design. At step 5980, bias values and primary and anysecondary negative loop inputs appropriate for this negative loop areobtained and the negative loop gain value calculated. The processadvances to step 5982 where the negative loop gain value is applied tothe combined output from loop 5974 to execute negative loop feedback.The process then advances to step 5984 where the results of step 5982are saved, then to step 5986 where any signal conditioning and delaycompensation is applied and saved in step 5988. The process then loopsback to 5973 and exits at step 5990 once the appropriate number of loopshave been completed at step 5973. Steps 5980, 5982, and 5986 areequivalent to blocks 5302B, 5305B-g, and 5310C-m of FIG. 52A.

[0486] Referring next to FIG. 60A, the processing of the loopcomparisons step 5815 of FIG. 58 can be better understood. The processstarts at step 6000, after which the process advances to step 6004 wherea loop is begun with the number of iterations of the loop being definedby how many serial linked positive and negative comparisons are to beprocessed. Serial linking allows later comparisons to use earliercomparisons processing typically signal processing such as lowpassfiltering. The number of serial levels of comparison can vary over awide range, and is identified here as simply 1 to I. When the processloop begins at step 6008, another loop is entered to process all of thenegative loop comparisons at this level, the number of comparisonidentified here as 1 to n. This process loop advances to step 6012 to dothe actual negative loop comparison after which the process loops backto step 6008. Once the appropriate number of loops have been completedat step 6008, the process advances to step 6016 where another loop isentered to process all of the positive loop comparisons at this level,the number of comparison identified here as 1 to p. This process loopadvances to step 6020 to do the actual positive loop comparison afterwhich the process loops back to step 6016. Once the appropriate numberof loops have been completed at step 6016, the process loops back tostep 6004. Once the appropriate number of loops have been completed atstep 6004, the process exits at step 6024.

[0487] Referring to FIG. 60B, the process of implementing negative loopcomparisons 6012 can be better understood. The process starts at step6030, after which the process advances to step 6034, where signalconditioning is performed on the appropriate processed environment powerestimator output, typically from loops 5902, 5908, 5916 or 5923 or apreviously computed environment negative value estimate from a previouscomparison level, to provide and save an environment negative valueestimate for this negative comparison loop.

[0488] The process loop advances to step 6038 where signal conditioningis performed on the appropriate processed reference power estimatoroutput, typically from loops 5952, 5958,5966 or 5973 or previouslycomputed reference positive value estimate from a previous comparisonlevel, to provide and save a reference positive value estimate for thisnegative comparison loop. The process advances to step 6042, where acheck is made for a closed loop noise compensation configuration. Iffalse, a leakage loop configuration has been selected, and the processadvances to step 6046, where the environment negative value estimate iscompared to the reference values, typically the same system referencepower estimator or previously computed reference positive value used instep 6038, to generate a negative loop gain offset value (in a mannersimilar to previously described block 5345 of FIG. 53D), and the resultsaved for use by the negative loop feedback, after which the processexits at step 6050. If true, closed loop noise compensation is to beperformed, and the process loop advances to step 6054, where A iscomputed as (reference positive value estimate—environment negativevalue estimate.) The process loop advances to step 6058 where the Avalues are compared to the reference values to generate a negative loopgain offset, that is saved for subsequent negative loop feedback use,after which the process exits at step 6050. Steps 6054 and 6058 can beperformed in a manner similar to previously described blocks 5315 and5345 of FIG. 53D.

[0489] Turning next to FIG. 60C, the processing of positive loopcomparisons 6020 can be better understood. The process starts at step6060, after which the process advances to step 6064 where signalconditioning is performed on the appropriate processed environment powerestimator output, typically from loops 5902, 5908, 5916 or 5923 or apreviously computed environment positive value estimate from a previouscomparison level, to provide and save the environment positive valueestimate for the current positive loop.

[0490] The process loop advances to step 6068 where signal conditioningis performed on the appropriate processed reference power estimatoroutput, typically from loops 5952, 5958, 5966 or 5973 or previouslycomputed reference value estimate from a previous comparison level, toprovide a reference negative value estimate for the current positiveloop. The process advances to step 6072 where a check is made for aclosed loop noise compensation configuration. If false, the processexits at step 6076 and the environment positive value estimate from step6064 is used later by noise processor 5820. If true, closed loop noisecompensation is to be performed, and the process advances to step 6080,where Δ=(environment positive value estimate−reference negative valueestimate) is computed. The process loop advances to step 6084, where acheck is made for Δ>0 since typically only positive values are used toindicate noise levels. If true, the process exits at step 6076. Iffalse, the process advances to step 6088, where Δ is set to 0 toindicate no noise, and the process exits at step 6076. Steps 6064, 6068,6080, 6088 can be performed in a manner similar to previously describedblocks 5310B, 5310D, 5350 and 5355 of FIG. 53A.

[0491] Referring next to FIG. 61A, the processing of the noise processor5820, previously described in connection with FIGS. 52C, and 54A-F, canbe better understood. The process starts at step 6100, after which theprocess advances to step 6102 where errors caused by the negative loopare removed. The process advances to step 6104 where the volume controlgain offset is calculated for designs that do not include a companderand use a volume control. The process then advances to variableattack/release step 6106 where the noise floor is processed to providethe correct response to the environmental noise and then to sensitivitycontrol step 6108 where the user can adjust the signal to noise ratio ofthe system. The process exits at step 6110. These four processing stepsare equivalent to the four noise processing sections described inconnection with FIG. 52C. Steps 6102 through 6108 may be executed in anyother order than the one shown and not all steps are required for everydesign. For example, corrections step 6102 need not be implemented ifthere are no negative loops to process and step 6104 need not beimplemented if the compander method of gain control is used.

[0492] Turning next to FIG. 61B, the processing of corrections 6102 canbe better understood. The process starts at step 6112, after which theprocess advances to step 6114 where a loop is begun with the number ofiterations of the loop being defined by how many positive loop outputsare to be processed. The number of positive loop outputs can vary over awide range, and is identified here as simply 0 to p, zero being used ifthere are no positive loop outputs to correct. When the process loopbegins at step 6116, the appropriate positive loop output and referencepower estimate is obtained, the differential error eliminated in step6118, and negative loop errors are corrected in step 6120. The processthen advances to step 6122 where numeric conversions may be processed.In some cases, computation requirements can be minimized by convertingthe results into an alternative value such as a logarithmic value. Theprocess then advances to step 6124 where the results of the previoussteps can be decimated or signal processing performed to reducecomputation requirements and increase loop stability typically by theuse of lowpass filtering. The process advances to step 6126, where theresults are saved for later use, and the process loops back to step6114. Once the appropriate number of loops have been completed at step6114, the process exits at step 6128. Not all processing steps arerequired for all designs. For example, the differential error may beinsignificant so step 6118 is skipped, there may not be a negative loopso step 6120 is skipped, or the numeric conversion 6122 ordecimation/signal processing 6124 may not be required.

[0493] Turning next to FIG. 61C, the volume control offset processing6104 can be better understood. The process starts at step 6130, afterwhich the process advances to step 6132 where a loop is begun with thenumber of iterations of the loop being defined by how many volumecontrol offsets are to be processed. The number of volume controloffsets to process can vary over a wide range, and is identified here assimply 0 to v, zero being used if there are no volume control offsets toprocess. When the process loop begins at step 6134, the appropriatepositive loop outputs, positive loop corrected outputs from loop 6114,attack/release values from loop 6162, or sensitivity outputs from loop6182 are obtained and then combined in step 6136. The process advancesto step 6138 where the appropriate reference power estimate is obtained,and then to step 6140 where the power estimate and combined value areused to calculate a volume control gain offset. The process thenadvances to step 6142 where numeric conversions may be processed. Insome cases, computation requirements can be minimized by converting theresults into an alternative value such as a logarithmic value. Theprocess then advances to step 6144 where the results of the previoussteps can be decimated or signal processing performed to reducecomputation requirements and increase loop stability typically by theuse of lowpass filtering. The process advances to step 6146, where theresults are saved for later use, and the process loops back to step6132. Once the appropriate number of loops have been completed at step6132, the process exits at step 6148. Not all processing steps arerequired for all designs. For example, the combiner 6136 is not requiredfor single inputs or the numeric conversion 6142 or decimation/signalprocessing 6144 may not be required.

[0494] Turning next to FIG. 61D, the variable attack/release processing6106 can be better understood. The process starts at step 6160, afterwhich the process advances to step 6162 where a loop is begun with thenumber of iterations of the loop being defined by the number of variableattack/release processes. The number of variable attack/releaseprocesses can vary over a wide range, and is identified here as simply 0to a, zero being used if there are no variable attack/release processes.When the process loop begins at step 6164, the appropriate positive loopoutputs, positive loop corrected outputs from loop 6114, volume controloffset values from loop 6132, or sensitivity outputs from loop 6182 areobtained and then combined in step 6165. The process advances to step6166 where the appropriate user interface inputs are obtained toconfigure the desired variable attack/release behavior and then to step6168 where the user interface inputs and combined value are applied tothe variable attack/release processor. The process then advances to step6170 where numeric conversions may be processed. In some cases,computation requirements can be minimized by converting the results intoan alternative value such as a logarithmic value. The process thenadvances to step 6172 where the results of the previous steps can bedecimated or signal processing performed to reduce computationrequirements and increase loop stability typically by the use of lowpassfiltering. The process advances to step 6174, where the results aresaved for later use, and the process loops back to step 6162. Once theappropriate number of loops have been completed at step 6162, theprocess exits at step 6176. Not all processing steps are required forall designs. For example, the combiner 6136 is not required for singleinputs or the numeric conversion 6142 or decimation/signal processing6144 may not be required.

[0495] Turning next to FIG. 61E, the sensitivity control processing 6108can be better understood. The process starts at step 6180, after whichthe process advances to step 6182 where a loop is begun with the numberof iterations of the loop being defined by the number of sensitivitycontrols. The number of sensitivity controls can vary over a wide range,and is identified here as simply 0 to s, zero being used if there are nosensitivity controls. When the process loop begins at step 6184, theappropriate positive loop outputs, positive loop corrected outputs fromloop 6114, volume control offset values from loop 6132, or variableattack/release outputs from loop 6162 are obtained and then combined instep 6186. The process advances to step 6188 where the appropriate userinterface inputs are obtained to set the desired system signal to noiseratio and then to step 6190 where the user interface inputs and combinedvalue are applied to the sensitivity control processor. The process thenadvances to step 6192 where numeric conversions may be processed. Insome cases, computation requirements can be minimized by converting theresults into an alternative value such as a logarithmic value. Theprocess then advances to step 6194 where the results of the previoussteps can be decimated or signal processing performed to reducecomputation requirements and increase loop stability typically by theuse of lowpass filtering. The process advances to step 6196, where theresults are saved for later use, and the process loops back to step6182. Once the appropriate number of loops have been completed at step6182, the process exits at step 6198. Not all processing steps arerequired for all designs. For example, the combiner 6136 is not requiredfor single inputs or the numeric conversion 6142 or decimation/signalprocessing 6144 may not be required.

[0496]FIG. 62 shows an exemplary embodiment of a two-stage acoustic loopbalance processor 5360 which provides both coarse and fine environmentalinput adjust and is typically used in conjunction with a codec with acoarsely adjustable programmable input amplifier. Referring to FIG. 51,a calibration signal is supplied as a reference signal 4310 and speakeroutput 480 during a period when there is minimal environmental noise.Negative loops are disabled, and the reference and microphone powerestimate signals are compared, typically by subtraction 6210. Themicrophone balance gain signal 5364 is adjusted up and down until thereference and microphone power estimate signals are approximately equal.If they remain equal for the proper closure time, as indicated by theclosure counter 6240, the loop closure is complete, and the microphonefine adjust 5362 and microphone balance gain 5364 constants are set. Nofurther changes are made to the outputs of the balancing processor.

[0497] The processor may make use of an active low-pass filter 6200,6205, and 6215, whose corner frequency is decreased as the loop closuretime increases to process the reference and microphone power difference6210. This provides fast initial gain adjust, as well as accurate finalgain adjustment.

[0498] The acoustic loop balance processor 5360 is initialized by thestart loop closure 6255 signal, that loads the KL down counter 6200 withan initial value. The KL 6200 value is limit checked and limited ifrequired by 6205 and supplied as the KL corner frequency coefficient tothe low pass filter 6215. Difference 6210 as a function of slowreference power estimator 5340 and slow microphone power estimator 5335is supplied as input to low pass filter step 6215. Update delay 6220 isa delay counter clocked by sample clock 2105 or a decimated clockfrequency. With each clock enable, it resets itself, causing KL downcounter 6200 to decrement and enables compare block 6235. The delaylimits the reaction time of the balancing circuit, providing stableoperation.

[0499] The output of filter 6215 is supplied to the >1.5 dB compare6235. The >1.5 dB compare 6235 provides three outputs, equal (within+1-1.5 dB), greater than 1.5 dB, and less than 1.5 dB. The greater than1.5 dB and less than 1.5 dB outputs cause increases and decreases in thecodec gain counter 6250, producing microphone balance gain 5364. The AIDconverters in many codecs contain a coarsely adjustable programmableinput amplifier (in this example adjustable in 1.5 dB steps although anyappropriate step size can be used) that can be used to scale the analogsignal prior to conversion. While it is preferable to perform coarseadjustment of the microphone using this amplifier since it enables themaximum A/D resolution, best signal to noise ratio, and lowest componentcount and costs, alternate variations such as a digitalamplifier/attenuator may also be used.

[0500] The equal output of compare 6235 clocks a closure counter 6240that was initially set to zero by start loop closure 6255, and is alsoreset by the greater than 1.5 dB or less than 1.5 dB outputs of 6235.The output of the closure counter is provided to comparator 6245 whichcompares it to the calibration counter limit, and produces a loopclosure done 6260 when the closure counter indicates that the balancingoperation has remained constant for the desired amount of time (i.e. noresets by the <>1.5 dB outputs; only “=” outputs), and thus thecalibration operation is complete.

[0501] The error to gain offset transform 6225, similar to previouslydescribed blocks 5205 and 5345, uses the slow reference power estimator5340 and filter output 6215 to generate a fine adjust gain signal. Atthe time that loop closure 6260 is asserted, this signal is accepted bythe register 6230 and becomes the microphone fine adjust signal 5362. Itallows adjustment of less than +/−1.5 dB to the balance loop.

[0502] Turning next to FIG. 63, the loop balance process 1230 can bebetter understood. The acoustic loop balancing processor 5360, anexample of which was shown in FIG. 62, represents an implementation ofthe flow diagram shown in FIG. 63.

[0503] The process starts at step 6300, after which the process advancesto step 6305 where a loop is begun with the iteration of the loop beingdefined by how many channels are to be processed. The number of channelsplus a final overall balance loop can vary over a wide range, and isidentified here as simply 1 to c. When the process loop begins at step6310, the negative and positive loops are disabled, the closure counteris initialized to zero, the balance gain is set to its maximum gain, andthe calibration 420 source is enabled.

[0504] The process loop advances to step 6315 where a calculation ofΔ=reference−microphone is made. The process loop advances to step 6320where the Δ is processed through a variable Fc low pass filter. Theprocess loop advances to step 6325 where a check is made ofΔ>+tolerance. In FIG. 62, this tolerance was 1.5 dB. If true, theprocess loop advances to step 6330 where the balance gain is increased,and then advances to step 6335, where an optional delay for loopstability gain change settling time is performed. The process loopadvances to step 6340 where the closure counter is reset, since thebalance loop is not yet balanced. The process loop advances to step6360, where a check is made for the counter >limit. If true, step 6365is executed, causing the residual Δ error to be used as the microphonefine adjustment gain value, and the process loops back to step 6305.

[0505] If the check at step 6325 was false, the process loop advances tostep 6345, where a check is made for Δ<+tolerance. If true the processloop advances to step 6350 where the balance gain is decreased. Theprocess loop advances to step 6335, and repeats the prescribed steps.

[0506] If the check at step 6345 was false, the process loop advances tostep 6355, where the closure counter is incremented. The process loopadvances to step 6360. If false, the process loop advances to step 6315.Once the appropriate number of loops have been completed at step 6305,the process exits at step 6370.

[0507] The following methods enable proper dynamic range mapping of thepartitioned signal processing system when volume control changes aremade or noise compensation is performed. Prior-art companders and/ornoise compensators can also use these methods. In prior art methods,when proper dynamic range mapping was initially achieved, subsequentvolume control changes destroyed it resulting in undesirable dynamicrange mapping.

[0508] In the simplest “set maximum and minimum” method, as shown inFIGS. 64 and 65, the user explicitly sets the maximum and minimumvolumes to set system gain and compander kneepoints (and associatedcompanding ratios) to accomplish dynamic range mapping.

[0509]FIG. 64 illustrates a configuration of the partitioned signalprocessing system for setting the maximum acoustic signal amplitude.First, User Interface 405 enters the “set maximum output level” setupmode via User Controls 310. User Interface 405 then enables Calibrator420 which provides a maximum amplitude signal, typically a 0 dB whitenoise signal, to the volume control 445. The Volume Control is set toits maximum volume level by User Interface 405. The volume controlsignal is sent to the Output Signal Processor 475. The user adjustsoutput gain “G” to set the maximum desired acoustic sound pressure level(SPL) by means of the user interface 405 and User Controls 310,typically the Volume Up/Down controls. User Interface 405 exits from“set maximum output level” setup mode via User Controls 310 with thesystem gain set to “G” and upper kneepoint typically set to apredetermined value. In this and following examples, the upper kneepointinput signal level is the same as the output signal level, both levelsbeing set to the maximum input signal level of the system although anyother input/output levels may be used. Note that the set maximum outputlevel procedure can be implemented on a per channel or band basis.

[0510] The configuration shown in FIG. 65 is used to set the minimumacoustic signal amplitude, typically at the user's threshold of hearing.First, User Interface 405 enters the “set minimum output level” setupmode via User Controls 310. User Interface 405 then enables Calibrator420 which provides a maximum amplitude signal, typically a 0 dB whitenoise signal, to the volume control 445. The Output Signal Processor 475output gain has previously been set to “G” from the “set maximum step”.The user adjusts volume control 445 by means of the user interface 405,and User Controls 310, typically the Volume Up/Down controls, to thepoint where the desired minimum SPL sound may be detected. UserInterface 405 exits from “set minimum output level” setup mode via UserControls 310 with the lower kneepoint being set. In this and followingexamples, the lower kneepoint input signal level is set to the minimuminput signal level (Oust above the noise floor) and the lower kneepointoutput signal level is set to volume control indicated level from the“set minimum output level” routine. At the completion of these two setmaximum/minimum steps, the dynamic range mapping has been set by thesetting the compander kneepoints and associated companding ratio andsystem gain. Note that the set minimum output level procedure can beimplemented on a per channel or band basis, such that the exemplarymethod described herein may be extended to any desired number of bandsor channels.

[0511] Once the dynamic range mapping has been set, it is desirable toavoid altering the minimum output setting when subsequent volume controlchanges are made so that the softest sounds can always be heard.Reducing the volume control reduces the overall system gain whichrequires the output dynamic range of the signal to be reduced, since theminimum volume should not be changed. This requires the signal to haveadditional compression, in addition to lower volume control. Theconverse is true for increasing the maximum volume. As with the priormethod, this method may be extend to any desired number of channels andbands of a system.

[0512]FIG. 66 illustrates how system gain or volume control changes canbe accomplished without destroying the minimum level. User Controls 310,typically the volume up/down controls, are received by User Interface405 which indicates to Transform Engine 410 the minimum output level andthe desired volume level. Transform Engine 405 then “transforms” thisinformation into both a volume control level provided to Volume Control445, which modifies the overall system gain, and compander operatingparameters, typically a modified lower kneepoint output signal level andassociated higher companding ratio, provided to Compander 450. In thisexample, a 10 dB decrease in the volume control results in a 10 dBincrease in compander compression so the minimum level remains the same.In this example, the Output Signal Processor output gain “G” was fixedduring the set maximum step however the volume control changes mayalternatively be implemented by changing output gain “G”.

[0513] Input Signal Preprocessing 440 is also shown to provide inputlevel matching for optimum compander operation.

[0514] The “set maximum and minimum” method is inappropriate for someusers because they may not know how loud the maximum volume will need tobe when it is noisy. The “set typical and minimum” method addresses thisconcern by allowing the user to set the typical maximum volume they wishto hear, while reserving adequate headroom to accommodate likely maximumvolume levels acceptable for noisy environments. The methods describedhere for a single band or channel may be expanded to any desiredarrangement of channels and bands.

[0515]FIG. 67 shows a “set typical maximum” level where the system gainheadroom is provided by a volume control bias, in this example −20 dB.The method is the same as described in FIG. 64, “set maximum”, exceptthat User Interface 405 provides to Volume Control 445 a volume levelwith bias instead of the maximum volume level.

[0516]FIG. 68 shows a “set typical maximum” level where the system gainheadroom is provided by an output gain bias, in this example −20 dB. Themethod is the same as described in FIG. 64, “set maximum”, except thatUser Interface 405 provides to Output Signal Processor 475 an outputgain value with bias.

[0517] The “set minimum” of the “set typical and minimum” method isperformed identically to FIG. 65.

[0518] The “automatic” method provides the most user-friendly interface,since it appears to be a normal volume control. This method uses defaultvalues, typically decided by design and set at power on or correspondingto the last use settings, and intelligence to determine how the outputdynamic range (user maximum and minimum output levels) should bealtered. When the signal is relatively loud, if the user adjusts thevolume, it is typically to alter the maximum output setting. If thesignal is relatively quiet, it is usually to modify the user minimumoutput setting. As before, this method may be applied to any desiredarrangement of multichannel and multiband systems.

[0519]FIG. 69 shows a typical design default setting example where thesystem gain output headroom is provided by a volume control bias.Default compander 450 and volume control 445 settings are set by UserInterface 405 and Transform Engine 410. Output Signal Processor 475output system gain G, is fixed by design. In this example, a 40 dBoutput dynamic range, with 80 dB maximum and 40 dB minimum SPL, and −20dB volume control bias is shown. An 80 dB input dynamic range is alsoshown being provided to Input Signal Preprocessing 440 which results ina default compander setting of 2:1 compression with the upper kneepointinput and output signal levels set to 0 dB and the lower kneepoint inputsignal level set to −80 dB and output signal level set to 40 dB.

[0520]FIG. 70 shows an example of how the maximum output level can beautomatically increased. If User Controls 310, typically a volumeup/down control, indicates to User Interface 405 to increase the outputvolume, a test is performed to determine which settings to modify. Thistest typically compares the current input power level, provided byCompander 450, to a threshold level, typically provided by a StatisticalEngine 415 or determined by design. If the input power level is greaterthan the threshold, an increase in the maximum output level is indicatedby User Interface 405. Transform Engine 410 receives inputs from theUser Interface which results in an increase in the Volume Control 445level (increase in system gain) and a decrease in Compander 450compression (lower companding ratio). Thus an automatic increase in themaximum output level, while maintaining the minimum output level, isaccomplished with one user volume control.

[0521]FIG. 71 shows an example of how the maximum output level can beautomatically decreased. If User Controls 310, typically a volumeup/down control, indicates to User Interface 405 to decrease the outputvolume, a test is performed to determine which settings to modify. Thistest typically compares the current input power level, provided byCompander 450, to a threshold level, typically provided by a StatisticalEngine 415 or determined by design. If the input power level is greaterthan the threshold, a decrease in the maximum output level is indicatedby User Interface 405. Transform Engine 410 receives inputs from theUser Interface which results in a decrease in the Volume Control 445level (system gain) and a increase in Compander 450 compression (highercompanding ratio). Thus an automatic decrease in the maximum outputlevel, while maintaining the minimum output level, is accomplished withone user volume control.

[0522]FIG. 72 shows an example of how the minimum output level can beautomatically increased. If User Controls 310, typically a volumeup/down control, indicates to User Interface 405 to increase the outputvolume, a test is performed to determine which settings to modify. Thistest typically compares the current input power level, provided byCompander 450, to a threshold level, typically provided by a StatisticalEngine 415 or determined by design. If the input power level is lessthan the threshold, an increase in the minimum output level is indicatedby User Interface 405. Transform Engine 410 receives inputs from theUser Interface which results in an increase in Compander 450 compression(higher companding ratio) and no modifications to the Volume Control 445level (system gain). Thus an automatic increase in the minimum outputlevel, while maintaining the maximum output level, is accomplished withone user volume control. Alternatively, an increase in the VolumeControl 445 level may accompany the increase in compander minimum levelto reduce the amount of compression and signal distortion.

[0523]FIG. 73 shows an example of how the minimum output level can beautomatically decreased. If User Controls 310, typically a volumeup/down control, indicates to User Interface 405 to decrease the outputvolume, a test is performed to determine which settings to modify. Thistest typically compares the current input power level, provided byCompander 450, to a threshold level, typically provided by a StatisticalEngine 415 or determined by design. If the input power level is lessthan the threshold, a decrease in the minimum output level is indicatedby User Interface 405. Transform Engine 410 receives inputs from theUser Interface which results in a decrease in Compander 450 compression(lower companding ratio) and no modifications to the Volume Control 445level (system gain). Thus an automatic decrease in the minimum outputlevel, while maintaining the maximum output level, is accomplished withone user volume control.

[0524]FIG. 74 is an alternative example of the default settings shown inFIG. 69. It shows how the same acoustic output levels can be obtained bymoving the system gain headroom from the volume control to the OutputSignal Processor output gain “G”. In this example, the Output SignalProcessor output gain is reduced by 20 dB, while the volume control gainis set to maximum by the Transform Engine 410.

[0525] In FIGS. 70, 71, 72, and 73, Output Signal Processor 475 outputgain may be used in addition to or instead of the VolumeControl/Pre-mixer 445 adjustments.

[0526] The “noise compensation with compander” method shown in FIG. 75shows how an increase in the noise floor determined by Noise Extractor465 results in Transform Engine 410 to increase the amount of Compander450 compression (higher companding ratio) so that the minimum outputlevel is greater than the noise floor. Alternatively, the system gain,via Volume Control 445 level and Output Signal Processor 475 outputgain, may be increased to help minimize the amount of compression, ormay be increased once a predetermined maximum compression level isreached. User Controls 310 may indicate to User Interface 405 toincrease or decrease the amount of Noise Sensitivity Control provided toNoise Extractor 465. This allows the user to manually adjust the minimumoutput level signal to noise ratio. The Noise Sensitivity Control mayalso be automatically adjusted by User Interface 405 as described byFIG. 78. As discussed previously, this method may be expanded tomultichannel and multiband systems.

[0527] The “noise compensation without compander” method shown in FIG.76 indicates how the partitioned signal processing system may be used toeffect a change in output volume when a noise floor signal is determinedby Noise Extractor 465, and Volume Control 445 is used without Compander450. As before, this method may be readily expanded to multichannel andmultiband systems.

[0528] In FIG. 76, the non-compander noise compensation compares thenoise floor plus noise sensitivity level to the current system outputlevel (system reference) and increases the system gain until the systemoutput is greater than or equal to the noise floor plus noisesensitivity level.

[0529] This is accomplished by a Channel Reference Out 4310 signal fromOutput Signal Processor 475 and Noise Sensitivity Control signal fromUser Interface 405 (same as described in connection with FIG. 75) beingsupplied to Noise Extractor 465, that in turn generates a Noise Offset5115 signal. The Transform Engine 410 uses the Noise Offset signal todetermine how to vary the Output Signal Processor 475 output gain “G”while not modifying the Volume Control 445 level. This allows the normalvolume control and noise compensation to be implemented in two differentmodules. Alternatively, the Volume Control and Output Signal Processoroutput gain may be operated in tandem or the Output Signal Processoroutput gain may be fixed and the volume increased via the volumecontrol.

[0530]FIG. 77 indicates in a flow diagram form the manual methodsdiscussed in FIGS. 64,65,67,68 regarding setting maximum and minimumoutput levels. FIG. 78 indicates in a flow diagram form the manualmethod discussed in FIGS. 75, 76 regarding how to modify the noisesensitivity control. These flow diagrams show a typical methodconsistent with a User Interface as described in FIG. 6C.

[0531] Referring next to FIG. 77, the set-up command decoder step 660Eand setup command execute step 660F shown in FIG. 6C may be betterunderstood through the conceptual description of an exemplaryimplementation of the “set maximum” and “set minimum” commands may beimplemented.

[0532] The process starts at step 7700, after which a check is made atstep 7705 to determine if a set maximum command is to be executed. Herea user will adjust the maximum output level of the system. If true, step7710 activates a calibration source 420, as shown in FIG. 64 and theUser Interface enters “set maximum output level” setup mode so thatcheck step 7705 will be true on subsequent passes through the set-upcommand decoder 660E and setup command execute 660F. The loop advancesto step 7715 where a check is made for an increase volume (volume+)command. If true, step 7720 is executed, increasing the output signalprocessor 475 output gain, as shown in FIG. 64. If false, a check ismade at step 7730 for a decrease volume (volume−) command. If true, step7735 decreases the output signal processor 475 output gain, as shown inFIG. 64. If neither a volume+or volume− command is made, no change ismade but the “set maximum output level” setup mode continues.

[0533] Next, the loop advances to step 7725 where a check is made as towhether to exit the set maximum command. If true, step 7740 causes thenoise extractor 465 to perform a loop balance operation to obtainoptimum noise compensation. Once this operation is complete, the loopadvances to step 7745 that disables the calibration source 420 and theUser Interface exits “set maximum output level” setup mode. The processthen exits at step 7795. If step 7725 is false, the User Interfaceremains in “set maximum output level” setup mode and proceeds to step7795.

[0534] Note that the “set maximum output level” setup mode may beexecuted many times, on a per channel and per band basis.

[0535] If the check made at step 7705 was false, the loop advances to acheck at step 7750 to determine if a set minimum command is to beexecuted. Here a user will, typically, adjust the output level of thecalibration source until it is just perceptible—which is typically theusers threshold of hearing. If true, step 7755 activates a calibrationsource 420, as shown in FIG. 64 and the User Interface enters “setminimum output level” setup mode so that check step 7750 will be true onsubsequent passes through the set-up command decoder 660E and setupcommand execute 660F. The loop advances to a check at step 7760 for avolume+ command. If true, the loop advances to step 7765, increasing thevolume control & pre-mixer 445, as shown in FIG. 65 which effectivelyincreases the calibration output level. If false, the loop advances tostep 7780, where a check is made for a volume− command. If true, step7785 is executed, decreasing the volume control & pre-mixer 445, asshown in FIG. 65, effectively decreasing the calibration output level.If neither a volume+or volume− command is made, no change is made butthe “set minimum output level” setup mode continues.

[0536] Next the loop advances to step 7770 where a check is made forexiting the set minimum command. If true, step 7775 is executed,disabling the calibration source 420, and the User Interface exits the“set minimum output level” setup mode. The process then exits at step7795. If step 7770 is false, the User Interface remains in the “setminimum output level” setup mode and proceeds to step 7795.

[0537] Note that the “set minimum output level” setup mode may beexecuted on a per channel and per band basis as many times as requiredfor multiband and multichannel implementations.

[0538] If the check at step 7750 was false, indicating that neither aset maximum or minimum level command was selected, step 7790 is executedto check if a sensitivity command is to be executed, as will bediscussed in connection with FIG. 78. Step 7793 is then executed,causing other commands (see discussion at FIG. 6A, above) to be decodedand executed, and the process exits at step 7795.

[0539] The noise sensitivity control allows the signal to noise ratio ofthe desired output signal relative to the noise floor to be controlled.By increasing the sensitivity, the sound is given priority over thenoise, and will be more easily heard over the noise level. By decreasingthe sensitivity, the environmental noise, typically speech, is givenpriority so that conversations will not be overpowered by an increasingsystem output level. The noise sensitivity control is typically appliedto all channels and bands in a system.

[0540] Referring to FIG. 78, the steps shown provide a conceptualdescription of how the setup sensitivity command may be implemented. Theprocess starts at step 7800. A check is made at step 7805 for asensitivity setup command. If true, a check is made at step 7810 as towhether pre-programmed sensitivity settings are to be used. This isuseful for systems that have discrete sensitivity settings, for example“music priority”, “conversation priority”, or “noise compensation off”settings. If check step 7810 is true, then step 7840 is executed wheretypically user controls are used to directly select which pre-programmedsetting to use in a “round robin” approach where a user control cyclesthrough various settings. Step 7825 is then executed which applies theselected sensitivity setting to the system followed by exiting at step7845.

[0541] If a user desires a continuum of settings, a user control,typically the volume control, may be used to increase or decrease thesensitivity level. This is the path selected when check step 7810 isfalse. If false, a check is made at 7815 for a volume+ command. If true,step 7820 causes the sensitivity to be increased. Step 7825 is thenexecuted causing the sensitivity settings to be applied before exitingat step 7845.

[0542] If the check at step 7815 was false, a check is made at step 7830for a volume− command. If true, the sensitivity is decreased at step7835, and applied at step 7825, before exiting at step 7845. If for somereason no increase or decrease in sensitivity was indicated, check step7830 will fail and no change in the noise sensitivity control value willresult.

[0543] Referring next to FIG. 79, there is shown therein in flow diagramform a Conceptual Intelligent Volume Control. In particular, FIG. 79shows how the automatic method of FIGS. 70,71,72,73 (to set the minimumand maximum output levels) and FIGS. 75,76 (to set the noise sensitivitycontrol level) may be implemented with operational command decoder 660Cand operation command 660D shown in FIG. 6C.

[0544] The process starts at step 7900, after which a check is made atstep 7905 for an increase/decrease volume control command. If false,step 7910 is executed to decode and execute any other commands, followedby the process exiting at step 7970. If the check at step 7905 is true,a check is made at step 7915 for a change sensitivity command. This isaccomplished typically by comparing the channel or system referencelevel to the noise floor. If the reference is close to the noise floorthen a change in the noise sensitivity control level is indicated andstep 7915 will be true. If true, a check is made at step 7920, for avolume+ command. If true, step 7925 increases the amount of noisesensitivity, as shown in FIG. 75, and the process exits at step 7970. Iffalse, step 7930 decreases the amount of noise sensitivity, and theprocess exits at step 7970.

[0545] If the check at step 7915 was false, a check is made at step 7935for a change in minimum output level. This is accomplished typically bycomparing the current input power level, provided by Compander 450, to athreshold level, typically provided by a Statistical Engine 415 ordetermined by design as shown in FIGS. 72,73. If the input power levelis less than the threshold, a modification of the minimum output levelis detected and check step 7935 will be true. If true, a check is madeat step 7940 for a volume+ command. If true, step 7945 is executed,causing minimum output level to be increased, as discussed in FIG. 72,and the process exits at step 7970. If false, step 7950 is executed,causing the minimum output level to be decreased, as discussed inconnection with FIG. 73, and the process exits at step 7970.

[0546] As noted above, the test at step 7935 typically compares thecurrent input power level, provided by Compander 450, to a thresholdlevel, typically provided by a Statistical Engine 415 or determined bydesign. In contrast to the above discussion, if the input power level isgreater than the threshold, an increase in the maximum output level isindicated by User Interface 405.

[0547] If the check at step 7935 was false, then an increase/decreasemaximum output level operation is to be performed. A check is made atstep 7955 for a volume+ command. If true, step 7960 increases the volumecontrol as discussed in FIG. 70, and the process exits at step 7970. Iffalse, step 7965 decreases the volume control as discussed in FIG. 71,and the process exits at step 7970.

[0548] Having fully described a preferred embodiment of the inventionand various alternatives, those skilled in the art will recognize, giventhe teachings herein, that numerous alternatives and equivalents existwhich do not depart from the invention. It is therefore intended thatthe invention not be limited by the foregoing description, but only bythe appended claims.

What is claimed is:
 1. An input level matching system comprising aninput for providing at least one signal, level matching logic formapping the at least one signal to a predetermined signal amplitude togenerate a modified signal, an output for providing the modified signal.2. The input level matching system of claim 1 further including signalprocessing logic for performing at least one of signal selection,filtering, or analog-to-digital conversion of the at least one inputsignal.
 3. The input level matching system of claim 1 wherein the levelmatching logic includes a gain cell.
 4. The input level matching systemof claim 1 wherein the output of the gain cell is a 0 db signal.
 5. Theinput level matching system of claim 3 wherein the gain cell operates toprovide digital amplification.
 6. The input level matching system ofclaim 3 wherein the gain cell operates to provide analog amplification.7. The input level matching system of claim 3 further including a clipsignal, clip detection logic responsive to the clip signal for comparingthe clip signal to a threshold signal for detecting signal clipping atthe output of the gain cell.
 8. The input level matching system of claim7 wherein the clip signal is a function of the input to the gain cell.9. The input level matching system of claim 7 wherein the clip signal isa function of the output of the gain cell.
 10. The input level matchingsystem of claim 8 wherein the output of the gain cell is a 0 dB signal,and the gain of the gain cell may be adjusted to provide a current gain.11. The input level matching system of claim 10 wherein the thresholdsignal is a function of a maximum 0 dB signal divided by the currentgain of the gain cell.
 12. The input level matching system of claim 9wherein the output of the gain cell is a 0 dB signal, and the gain ofthe gain cell may be adjusted to provide a current gain.
 13. The inputlevel matching system of claim 12 wherein the clip signal is compared toa maximum 0 dB signal.
 14. The input level matching system of claim 11further including direct calculation logic for detecting when the clipdetector detects a clipping condition and directly calculating a maximumpermissible gain value.
 15. The input level matching system of claim 11further including gain reducing logic for reducing the current gain ofthe gain cell when the clip detector detects a clipping condition. 16.The input level matching system of claim 15 wherein the gain reducinglogic is adjusted iteratively to reduce the gain to an acceptable level.17. The input level matching system of claim 13 further including directcalculation logic for detecting when the clip detector detects aclipping condition and directly calculating a maximum permissible gainvalue.
 18. The input level matching system of claim 13 further includinggain reducing logic for reducing the current gain of the gain cell whenthe clip detector detects a clipping condition.
 19. The input levelmatching system of claim 18 wherein the gain reducing logic is adjustediteratively to reduce the gain to an acceptable level.
 20. The inputlevel matching system of claim 7 further including a clip detectorcounter.
 21. The input level matching system of claim 20 furtherincluding signaling logic for indicating the occurrence of a clipcondition.
 22. The input level matching system of claim 1 furtherincluding bandsplit and equalization logic.
 23. The input level matchingsystem of claim 22 wherein equalization comprises scaling.
 24. The inputlevel matching system of claim 3 further including gain cell gainstorage and wherein the gain cell provides a gain value such that atleast one gain value can be stored in and retrieved from the gainstorage.
 25. The input level matching system of claim 24 wherein the atleast one input signal comprises multiple input signals.
 26. The inputlevel matching system of claim 24 wherein a gain value can be stored foreach of the multiple input signals.
 27. The input level matching systemof claim 26 wherein a plurality of gain values can be stored andretrieved for each of the multiple input signals.
 28. The input levelmatching system of claim 25 wherein the multiple input signals arearranged in groups.
 29. The input level matching system of claim 25wherein the gain value for all input signals within a group is the same.30. The input level matching system of claim 29 wherein the gain valuefor a group is determined by the maximum input signal within that group.